School of Computing Science Simon Fraser University

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Transport Layer 3-1 School of Computing Science Simon Fraser University CMPT 371: Data Communications and CMPT 371: Data Communications and Networking Networking Chapter 3: Transport Layer Chapter 3: Transport Layer

description

School of Computing Science Simon Fraser University. CMPT 371: Data Communications and Networking Chapter 3: Transport Layer. Our goals: understand principles behind transport layer services: multiplexing/demultiplexing reliable data transfer flow control congestion control - PowerPoint PPT Presentation

Transcript of School of Computing Science Simon Fraser University

Page 1: School of Computing Science  Simon Fraser University

Transport Layer 3-1

School of Computing Science Simon Fraser University

CMPT 371 Data Communications and CMPT 371 Data Communications and NetworkingNetworking

Chapter 3 Transport LayerChapter 3 Transport Layer

Transport Layer 3-2

Chapter 3 Transport LayerOur goals understand principles behind transport layer

services multiplexingdemultiplexing reliable data transfer flow control congestion control

learn about transport layer protocols in the Internet UDP TCP

Transport Layer 3-3

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-4

Transport services and protocols provide logical

communication between app processes running on different hosts

transport protocols run in end systems send side breaks app

messages into segments passes to network layer

rcv side reassembles segments into messages passes to app layer

more than one transport protocol available to apps Internet TCP and UDP

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-5

Transport vs network layer

network layer logical communication between hosts

transport layer logical communication between processes relies on enhances

network layer services

Household analogy12 kids sending letters

to 12 kids processes = kids app messages =

letters in envelopes hosts = houses transport protocol =

Ann and Bill network-layer protocol

= postal service

Transport Layer 3-6

Internet transport-layer protocols reliable in-order

delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP

services not available delay guarantees bandwidth guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-8

Multiplexingdemultiplexing

application

transport

network

link

physical

P1 application

transport

network

link

physical

application

transport

network

link

physical

P2P3 P4P1

host 1 host 2host 3

= process= socket

delivering received segmentsto correct socket

Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)

Multiplexing at send host

Transport Layer 3-9

How demultiplexing works host receives IP datagrams

each datagram has source IP address destination IP address

each datagram carries 1 transport-layer segment

each segment has source destination port number

host uses IP addresses amp port numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(message)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 2: School of Computing Science  Simon Fraser University

Transport Layer 3-2

Chapter 3 Transport LayerOur goals understand principles behind transport layer

services multiplexingdemultiplexing reliable data transfer flow control congestion control

learn about transport layer protocols in the Internet UDP TCP

Transport Layer 3-3

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-4

Transport services and protocols provide logical

communication between app processes running on different hosts

transport protocols run in end systems send side breaks app

messages into segments passes to network layer

rcv side reassembles segments into messages passes to app layer

more than one transport protocol available to apps Internet TCP and UDP

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-5

Transport vs network layer

network layer logical communication between hosts

transport layer logical communication between processes relies on enhances

network layer services

Household analogy12 kids sending letters

to 12 kids processes = kids app messages =

letters in envelopes hosts = houses transport protocol =

Ann and Bill network-layer protocol

= postal service

Transport Layer 3-6

Internet transport-layer protocols reliable in-order

delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP

services not available delay guarantees bandwidth guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-8

Multiplexingdemultiplexing

application

transport

network

link

physical

P1 application

transport

network

link

physical

application

transport

network

link

physical

P2P3 P4P1

host 1 host 2host 3

= process= socket

delivering received segmentsto correct socket

Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)

Multiplexing at send host

Transport Layer 3-9

How demultiplexing works host receives IP datagrams

each datagram has source IP address destination IP address

each datagram carries 1 transport-layer segment

each segment has source destination port number

host uses IP addresses amp port numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(message)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 3: School of Computing Science  Simon Fraser University

Transport Layer 3-3

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-4

Transport services and protocols provide logical

communication between app processes running on different hosts

transport protocols run in end systems send side breaks app

messages into segments passes to network layer

rcv side reassembles segments into messages passes to app layer

more than one transport protocol available to apps Internet TCP and UDP

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-5

Transport vs network layer

network layer logical communication between hosts

transport layer logical communication between processes relies on enhances

network layer services

Household analogy12 kids sending letters

to 12 kids processes = kids app messages =

letters in envelopes hosts = houses transport protocol =

Ann and Bill network-layer protocol

= postal service

Transport Layer 3-6

Internet transport-layer protocols reliable in-order

delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP

services not available delay guarantees bandwidth guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-8

Multiplexingdemultiplexing

application

transport

network

link

physical

P1 application

transport

network

link

physical

application

transport

network

link

physical

P2P3 P4P1

host 1 host 2host 3

= process= socket

delivering received segmentsto correct socket

Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)

Multiplexing at send host

Transport Layer 3-9

How demultiplexing works host receives IP datagrams

each datagram has source IP address destination IP address

each datagram carries 1 transport-layer segment

each segment has source destination port number

host uses IP addresses amp port numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(message)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 4: School of Computing Science  Simon Fraser University

Transport Layer 3-4

Transport services and protocols provide logical

communication between app processes running on different hosts

transport protocols run in end systems send side breaks app

messages into segments passes to network layer

rcv side reassembles segments into messages passes to app layer

more than one transport protocol available to apps Internet TCP and UDP

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-5

Transport vs network layer

network layer logical communication between hosts

transport layer logical communication between processes relies on enhances

network layer services

Household analogy12 kids sending letters

to 12 kids processes = kids app messages =

letters in envelopes hosts = houses transport protocol =

Ann and Bill network-layer protocol

= postal service

Transport Layer 3-6

Internet transport-layer protocols reliable in-order

delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP

services not available delay guarantees bandwidth guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-8

Multiplexingdemultiplexing

application

transport

network

link

physical

P1 application

transport

network

link

physical

application

transport

network

link

physical

P2P3 P4P1

host 1 host 2host 3

= process= socket

delivering received segmentsto correct socket

Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)

Multiplexing at send host

Transport Layer 3-9

How demultiplexing works host receives IP datagrams

each datagram has source IP address destination IP address

each datagram carries 1 transport-layer segment

each segment has source destination port number

host uses IP addresses amp port numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(message)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 5: School of Computing Science  Simon Fraser University

Transport Layer 3-5

Transport vs network layer

network layer logical communication between hosts

transport layer logical communication between processes relies on enhances

network layer services

Household analogy12 kids sending letters

to 12 kids processes = kids app messages =

letters in envelopes hosts = houses transport protocol =

Ann and Bill network-layer protocol

= postal service

Transport Layer 3-6

Internet transport-layer protocols reliable in-order

delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP

services not available delay guarantees bandwidth guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-8

Multiplexingdemultiplexing

application

transport

network

link

physical

P1 application

transport

network

link

physical

application

transport

network

link

physical

P2P3 P4P1

host 1 host 2host 3

= process= socket

delivering received segmentsto correct socket

Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)

Multiplexing at send host

Transport Layer 3-9

How demultiplexing works host receives IP datagrams

each datagram has source IP address destination IP address

each datagram carries 1 transport-layer segment

each segment has source destination port number

host uses IP addresses amp port numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(message)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 6: School of Computing Science  Simon Fraser University

Transport Layer 3-6

Internet transport-layer protocols reliable in-order

delivery (TCP) congestion control flow control connection setup

unreliable unordered delivery UDP no-frills extension of

ldquobest-effortrdquo IP

services not available delay guarantees bandwidth guarantees

application

transportnetworkdata linkphysical

application

transportnetworkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysical

networkdata linkphysicalnetwork

data linkphysical

logical end-end transport

Transport Layer 3-7

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-8

Multiplexingdemultiplexing

application

transport

network

link

physical

P1 application

transport

network

link

physical

application

transport

network

link

physical

P2P3 P4P1

host 1 host 2host 3

= process= socket

delivering received segmentsto correct socket

Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)

Multiplexing at send host

Transport Layer 3-9

How demultiplexing works host receives IP datagrams

each datagram has source IP address destination IP address

each datagram carries 1 transport-layer segment

each segment has source destination port number

host uses IP addresses amp port numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(message)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 7: School of Computing Science  Simon Fraser University

Transport Layer 3-7

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-8

Multiplexingdemultiplexing

application

transport

network

link

physical

P1 application

transport

network

link

physical

application

transport

network

link

physical

P2P3 P4P1

host 1 host 2host 3

= process= socket

delivering received segmentsto correct socket

Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)

Multiplexing at send host

Transport Layer 3-9

How demultiplexing works host receives IP datagrams

each datagram has source IP address destination IP address

each datagram carries 1 transport-layer segment

each segment has source destination port number

host uses IP addresses amp port numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(message)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 8: School of Computing Science  Simon Fraser University

Transport Layer 3-8

Multiplexingdemultiplexing

application

transport

network

link

physical

P1 application

transport

network

link

physical

application

transport

network

link

physical

P2P3 P4P1

host 1 host 2host 3

= process= socket

delivering received segmentsto correct socket

Demultiplexing at rcv hostgathering data from multiplesockets enveloping data with header (later used for demultiplexing)

Multiplexing at send host

Transport Layer 3-9

How demultiplexing works host receives IP datagrams

each datagram has source IP address destination IP address

each datagram carries 1 transport-layer segment

each segment has source destination port number

host uses IP addresses amp port numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(message)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 9: School of Computing Science  Simon Fraser University

Transport Layer 3-9

How demultiplexing works host receives IP datagrams

each datagram has source IP address destination IP address

each datagram carries 1 transport-layer segment

each segment has source destination port number

host uses IP addresses amp port numbers to direct segment to appropriate socket

source port dest port

32 bits

applicationdata

(message)

other header fields

TCPUDP segment format

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 10: School of Computing Science  Simon Fraser University

Transport Layer 3-10

Connectionless demuxDatagramSocket serverSocket = new DatagramSocket(6428)

ClientIPB

P2

client IP A

P1P1P3

serverIP C

SP 6428

DP 9157

SP 6428

DP 5775

SP 5775

DP 6428SP 9157

DP 6428

UDP socket identified by (dst IP dst Port) datagrams with different src IPs andor src ports are directed to same socket

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 11: School of Computing Science  Simon Fraser University

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple source IP address source port number dest IP address dest port number

recv host uses all four values to direct segment to appropriate socket

Server host may support many simultaneous TCP sockets each socket identified

by its own 4-tuple

Web servers have different sockets for each connecting client non-persistent HTTP will

have different socket for each request

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 12: School of Computing Science  Simon Fraser University

Transport Layer 3-12

Connection-oriented demux (cont)

ClientIPB

P1

client IP A

P1P2P4

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P5 P6 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 13: School of Computing Science  Simon Fraser University

Transport Layer 3-13

Connection-oriented demux Threaded Web Server

ClientIPB

P1

client IP A

P1P2

serverIP C

SP 9157

DP 80

SP 9157

DP 80

P4 P3

D-IPCS-IP A

D-IPC

S-IP B

SP 5775

DP 80

D-IPCS-IP B

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 14: School of Computing Science  Simon Fraser University

Transport Layer 3-14

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 15: School of Computing Science  Simon Fraser University

Transport Layer 3-15

UDP User Datagram Protocol [RFC 768]

ldquono frillsrdquo ldquobare bonesrdquo Internet transport protocol

ldquobest effortrdquo service UDP segments may be lost delivered out of order

to app connectionless

no handshaking between UDP sender receiver

each UDP segment handled independently of others

Why is there a UDP no connection

establishment (which can add delay)

simple no connection state at sender receiver

small segment header no congestion control

UDP can blast away as fast as desired

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 16: School of Computing Science  Simon Fraser University

Transport Layer 3-16

UDP more

often used for streaming multimedia apps loss tolerant rate sensitive

other UDP uses DNS SNMP

reliable transfer over UDP add reliability at application layer application-specific

error recovery

source port dest port

32 bits

Applicationdata

(message)

UDP segment format

length checksumLength in

bytes of UDPsegmentincluding

header

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 17: School of Computing Science  Simon Fraser University

Transport Layer 3-17

UDP checksum

Sender treat segment contents

as sequence of 16-bit integers

checksum addition (1rsquos complement sum) of segment contents

sender puts checksum value into UDP checksum field

Receiver compute checksum of

received segment check if computed checksum

equals checksum field value NO - error detected YES - no error detected

But maybe errors nonetheless More later hellip

Goal detect ldquoerrorsrdquo (eg flipped bits) in transmitted segment

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 18: School of Computing Science  Simon Fraser University

Transport Layer 3-18

Internet Checksum Example Note

When adding numbers a carryout from the most significant bit needs to be added to the result

Example add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 19: School of Computing Science  Simon Fraser University

Transport Layer 3-19

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 20: School of Computing Science  Simon Fraser University

Transport Layer 3-20

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 21: School of Computing Science  Simon Fraser University

Transport Layer 3-21

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 22: School of Computing Science  Simon Fraser University

Transport Layer 3-22

Principles of Reliable data transfer important in app transport link layers top-10 list of important networking topics

characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 23: School of Computing Science  Simon Fraser University

Transport Layer 3-23

Reliable data transfer getting started

sendside

receiveside

rdt_send() called from above (eg by app) Passed data to deliver to receiver upper layer

udt_send() called by rdtto transfer packet over unreliable channel to

receiver

rdt_rcv() called when packet arrives on rcv-side of channel

deliver_data() called by rdt to deliver data to

upper

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 24: School of Computing Science  Simon Fraser University

Transport Layer 3-24

Reliable data transfer getting startedWersquoll incrementally develop sender receiver

sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer

but control info will flow on both directions

use finite state machines (FSM) to specify sender receiver

state1

state2

event causing state transitionactions taken on state transition

state when in this ldquostaterdquo next state

uniquely determined by next event

eventactions

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 25: School of Computing Science  Simon Fraser University

Transport Layer 3-25

Rdt10 reliable transfer over a reliable channel

underlying channel perfectly reliable no bit errors no loss of packets

separate FSMs for sender receiver sender sends data into underlying channel receiver read data from underlying channel

Wait for call from above packet = make_pkt(data)

udt_send(packet)

rdt_send(data)

extract (packetdata)deliver_data(data)

Wait for call from

below

rdt_rcv(packet)

sender receiver

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 26: School of Computing Science  Simon Fraser University

Transport Layer 3-26

rdt20 channel with bit errors

underlying channel may flip bits in packet checksum to detect bit errors

the question how to recover from errors acknowledgements (ACKs) receiver explicitly tells

sender that pkt received OK negative acknowledgements (NAKs) receiver

explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK

new mechanisms in rdt20 (beyond rdt10) error detection receiver feedback control msgs (ACKNAK) rcvr-

gtsender

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 27: School of Computing Science  Simon Fraser University

Transport Layer 3-27

rdt20

Wait for call from above

sndpkt = make_pkt(data checksum)udt_send(sndpkt)

extract(rcvpktdata)deliver_data(data)udt_send(ACK)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

rdt_rcv(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp isNAK(rcvpkt)

udt_send(NAK)

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Wait for ACK or

NAK

Wait for call from

below

rdt_send(data)

Stop-and-wait protocolSender sends one packet then waits for receiver response

sender

receiver

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 28: School of Computing Science  Simon Fraser University

Transport Layer 3-28

rdt20 has a fatal flaw

What happens if ACKNAK corrupted

sender doesnrsquot know what happened at receiver

Canrsquot we just retransmit NO possible duplicate

receiver wonrsquot know whether it is a new or re-transmitted pkt

Handling duplicates sender adds sequence

number to each pkt sender retransmits

current pkt if ACKNAK garbled

receiver discards (doesnrsquot deliver up) duplicate pkt

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 29: School of Computing Science  Simon Fraser University

Transport Layer 3-29

rdt21 sender handles garbled ACKNAKs

Wait for call 0 from

above

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)

rdt_send(data)

Wait for ACK or NAK 0 udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isNAK(rcvpkt) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt)

Wait for call 1 from

above

Wait for ACK or NAK 1

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 30: School of Computing Science  Simon Fraser University

Transport Layer 3-30

rdt21 receiver handles garbled ACKNAKs

Wait for 0 from below

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq0(rcvpkt)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq1(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

Wait for 1 from below

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp has_seq0(rcvpkt)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

rdt_rcv(rcvpkt) ampamp not corrupt(rcvpkt) ampamp has_seq1(rcvpkt)

rdt_rcv(rcvpkt) ampamp (corrupt(rcvpkt)

sndpkt = make_pkt(ACK chksum)udt_send(sndpkt)

sndpkt = make_pkt(NAK chksum)udt_send(sndpkt)

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 31: School of Computing Science  Simon Fraser University

Transport Layer 3-31

rdt21 discussion

Sender seq added to pkt two seq rsquos (01)

will suffice Why must check if

received ACKNAK corrupted

twice as many states state must

ldquorememberrdquo whether ldquocurrentrdquo pkt has 0 or 1 seq

Receiver must check if

received packet is duplicate state indicates

whether 0 or 1 is expected pkt seq

note receiver can not know if its last ACKNAK received OK at sender

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 32: School of Computing Science  Simon Fraser University

Transport Layer 3-32

rdt22 a NAK-free protocol same functionality as rdt21 using ACKs only

instead of NAK receiver sends ACK for last pkt received OK receiver must explicitly include seq of pkt being

ACKed

duplicate ACK at sender results in same action as NAK retransmit current pkt

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 33: School of Computing Science  Simon Fraser University

Transport Layer 3-33

rdt30 channels with errors and loss

Packets (data or ACKs) can be lost checksum seq ACKs retransmissions will be of help

but not enough

How do we handle lost packets sender waits ldquoreasonablerdquo amount of time for ACK

requires countdown timer

retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost)

retransmission will be duplicate but use of seq rsquos already handles this

receiver must specify seq of pkt being ACKed

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 34: School of Computing Science  Simon Fraser University

Transport Layer 3-34

rdt30 sender

sndpkt = make_pkt(0 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

Wait for

ACK0

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt1) )

Wait for call 1 from

above

sndpkt = make_pkt(1 data checksum)udt_send(sndpkt)start_timer

rdt_send(data)

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt0)

rdt_rcv(rcvpkt) ampamp ( corrupt(rcvpkt) ||isACK(rcvpkt0) )

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt) ampamp isACK(rcvpkt1)

stop_timerstop_timer

udt_send(sndpkt)start_timer

timeout

udt_send(sndpkt)start_timer

timeout

rdt_rcv(rcvpkt)

Wait for call 0from

above

Wait for

ACK1

rdt_rcv(rcvpkt)

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 35: School of Computing Science  Simon Fraser University

Transport Layer 3-35

rdt30 in action

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 36: School of Computing Science  Simon Fraser University

Transport Layer 3-36

rdt30 in action

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 37: School of Computing Science  Simon Fraser University

Transport Layer 3-37

Performance of rdt30

first packet bit transmitted t = 0

sender receiver

RTT

last packet bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

U sender

= L R

RTT + L R

U sender utilization ndash fraction of time sender busy sending

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 38: School of Computing Science  Simon Fraser University

Transport Layer 3-38

Performance of rdt30 example

example 1 Gbps link 15 ms e-e prop delay 1KB packet

U sender

= 008

30008 = 000027

microseconds

L R

RTT + L R =

1KB pkt every 30 msec -gt 33kBsec (= 0264 Mbps) throughput over 1 Gbps link

Very poor performance for high-speed links network protocol limits use of physical resources

Suggestions

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 39: School of Computing Science  Simon Fraser University

Transport Layer 3-39

Pipelined protocols

Pipelining sender allows multiple ldquoin-flightrdquo yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender andor receiver

Two generic forms of pipelined protocols go-Back-N selective repeat

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 40: School of Computing Science  Simon Fraser University

Transport Layer 3-40

Pipelining increased utilization

first packet bit transmitted t = 0

sender receiver

RTT

last bit transmitted t = L R

first packet bit arriveslast packet bit arrives send ACK

ACK arrives send next packet t = RTT + L R

last bit of 2nd packet arrives send ACKlast bit of 3rd packet arrives send ACK

U sender

= 024

30008 = 00008

microseconds

3 L R

RTT + L R =

Increase utilizationby a factor of 3

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 41: School of Computing Science  Simon Fraser University

Transport Layer 3-41

Go-Back-NSender k-bit seq in pkt header ldquowindowrdquo of up to N consecutive unackrsquoed pkts allowed

ACK(n) ACKs all pkts up to including seq n - ldquocumulative ACKrdquo may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n) retransmit pkt n and all higher seq pkts in window

ie go back to n

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 42: School of Computing Science  Simon Fraser University

Transport Layer 3-42

GBN sender extended FSM

Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])hellipudt_send(sndpkt[nextseqnum-1])

timeout

rdt_send(data)

if (nextseqnum lt base+N) sndpkt[nextseqnum] = make_pkt(nextseqnumdatachksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ else refuse_data(data)

base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer

rdt_rcv(rcvpkt) ampamp notcorrupt(rcvpkt)

base=1nextseqnum=1

rdt_rcv(rcvpkt) ampamp corrupt(rcvpkt)

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 43: School of Computing Science  Simon Fraser University

Transport Layer 3-43

GBN receiver extended FSM

ACK-only always send ACK for correctly-received pkt with highest in-order seq may generate duplicate ACKs need only remember expectedseqnum

out-of-order pkt discard (donrsquot buffer) -gt no receiver buffering simpler Re-ACK pkt with highest in-order seq (duplicate ACK)

Wait

udt_send(sndpkt)

default

rdt_rcv(rcvpkt) ampamp notcurrupt(rcvpkt) ampamp hasseqnum(rcvpktexpectedseqnum)

extract(rcvpktdata)deliver_data(data)sndpkt = make_pkt(expectedseqnumACKchksum)udt_send(sndpkt)expectedseqnum++

expectedseqnum=1sndpkt = make_pkt(expectedseqnumACKchksum)

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 44: School of Computing Science  Simon Fraser University

Transport Layer 3-44

GBN inaction

Window size N = 4

Go back to 2

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 45: School of Computing Science  Simon Fraser University

Transport Layer 3-45

Go-Back-N Do you see potential problems with GBN

Consider high-speed links with long delays (called large bandwidth-delay product pipes)

GBN can fill that pipe by having large N many unACKed pkts could be in the pipe A single lost pkt could cause a re-transmission

of a huge number (up to N) of pkts waste of bandwidth

Solutions

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 46: School of Computing Science  Simon Fraser University

Transport Layer 3-46

Selective Repeat

receiver individually acknowledges all correctly received pkts buffers pkts as needed for eventual in-order

delivery to upper layer

sender only resends pkts for which ACK not received sender timer for each unACKed pkt

sender window N consecutive seq rsquos again limits seq s of sent unACKed pkts

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 47: School of Computing Science  Simon Fraser University

Transport Layer 3-47

Selective repeat sender receiver windows

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 48: School of Computing Science  Simon Fraser University

Transport Layer 3-48

Selective repeat

data from above if next available seq in

window send pkt

timeout(n) resend pkt n restart

timer

ACK(n) in [sendbase sendbase+N-1]

mark pkt n as received if n == sendbase

advance window base to next unACKed seq

senderpkt n in [rcvbase rcvbase+N-

1]

send ACK(n) out-of-order buffer in-order deliver (also

deliver buffered in-order pkts) advance window to next not-yet-received pkt

pkt n in [rcvbase-Nrcvbase-1]

ACK(n)

otherwise ignore

receiver

Why does receiver ACK pkts in the previous window

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 49: School of Computing Science  Simon Fraser University

Transport Layer 3-49

Selective repeat in action

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 50: School of Computing Science  Simon Fraser University

Transport Layer 3-50

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 51: School of Computing Science  Simon Fraser University

Transport Layer 3-51

TCP Overview RFCs 793 1122 1323 2018 2581

full duplex data bi-directional data flow

in same connection MSS maximum

segment size

connection-oriented handshaking (exchange

of control msgs) initrsquos sender receiver state before data exchange

flow controlled sender will not

overwhelm receiver

point-to-point one sender one

receiver

reliable in-order byte steam no ldquomessage

boundariesrdquo

pipelined TCP congestion and flow

control set window size

send amp receive bufferssocketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 52: School of Computing Science  Simon Fraser University

Transport Layer 3-52

TCP segment structure

source port dest port

32 bits

applicationdata

(variable length)

sequence number

acknowledgement numberReceive window

Urg data pnterchecksum

FSRPAUheadlen

notused

Options (variable length)

URG urgent data (generally not used)

ACK ACK valid

PSH push data now(generally not used)

RST SYN FINconnection estab(setup teardown

commands)

bytes rcvr willingto accept

countingby bytes of data(not segments)

Internetchecksum

(as in UDP)

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 53: School of Computing Science  Simon Fraser University

Transport Layer 3-53

TCP seq rsquos and ACKsSeq rsquos

byte stream ldquonumberrdquo of first byte in segmentrsquos data

ACKs seq of next byte

expected from other side

cumulative ACKQ how receiver handles out-

of-order segments

A TCP spec doesnrsquot say up to implementer

Host A Host B

Seq=42 ACK=79 data = lsquoCrsquo

Seq=79 ACK=43 data = lsquoCrsquo

Seq=43 ACK=80

Usertypes

lsquoCrsquo

host ACKsreceipt

of echoedlsquoCrsquo

host ACKsreceipt of

lsquoCrsquo echoesback lsquoCrsquo

timesimple telnet

scenario

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 54: School of Computing Science  Simon Fraser University

Transport Layer 3-54

TCP reliable data transfer

TCP creates rdt service on top of IPrsquos unreliable service

Pipelined segments Cumulative acks TCP uses single

retransmission timer

Retransmissions are triggered by timeout events duplicate acks

Initially consider simplified TCP sender ignore duplicate acks ignore flow control

congestion control

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 55: School of Computing Science  Simon Fraser University

Transport Layer 3-55

TCP sender eventsdata rcvd from app Create segment with

seq seq is byte-stream

number of first data byte in segment

start timer if not already running (think of timer as for oldest unacked segment)

expiration interval TimeOutInterval

timeout retransmit segment

that caused timeout restart timer Ack rcvd If acknowledges

previously unacked segments update what is known

to be acked start timer if there are

outstanding segments

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 56: School of Computing Science  Simon Fraser University

Transport Layer 3-56

TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) switch(event)

event data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer

end of loop forever

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 57: School of Computing Science  Simon Fraser University

Transport Layer 3-57

SendBase= 120

TCP retransmission scenarios

Host A

Seq=100 20 bytes data

ACK=100

time

premature timeout

Host B

Seq=92 8 bytes data

ACK=120

Seq=92 8 bytes data

Seq=

92

tim

eout

ACK=120

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

lost ACK scenario

Host B

X

Seq=92 8 bytes data

ACK=100

time

Seq=

92

tim

eout

SendBase= 100

SendBase= 120

Sendbase= 100

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 58: School of Computing Science  Simon Fraser University

Transport Layer 3-58

TCP retransmission scenarios (more)

Host A

Seq=92 8 bytes data

ACK=100

loss

tim

eout

Cumulative ACK scenario

Host B

X

Seq=100 20 bytes data

ACK=120

time

SendBase= 120

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 59: School of Computing Science  Simon Fraser University

Transport Layer 3-59

TCP Round Trip Time and Timeout If TCP timeout is

too short premature timeout unnecessary retransmissions

too long slow reaction to segment loss

Q how to set TCP timeout value Based on RTT but RTT itself varies with time

We need to estimate current RTT

RTT Estimation SampleRTT measured time from segment transmission until

ACK receipt ignore retransmissions SampleRTT will vary want estimated RTT ldquosmootherrdquo average several recent measurements not just current SampleRTT

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 60: School of Computing Science  Simon Fraser University

Transport Layer 3-60

TCP Round Trip Time and TimeoutEstimatedRTT = (1- )EstimatedRTT + SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value = 0125

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 61: School of Computing Science  Simon Fraser University

Transport Layer 3-61

Example RTT estimationRTT gaiacsumassedu to fantasiaeurecomfr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 62: School of Computing Science  Simon Fraser University

Transport Layer 3-62

TCP Round Trip Time and TimeoutSetting the timeout EstimtedRTT plus safety margin

large variation in EstimatedRTT -gt larger safety margin first estimate of how much SampleRTT deviates from

EstimatedRTT

TimeoutInterval = EstimatedRTT + 4DevRTT

DevRTT = (1-)DevRTT + |SampleRTT - EstimatedRTT|

(typically = 025)

Then set timeout interval

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 63: School of Computing Science  Simon Fraser University

Transport Layer 3-63

Fast Retransmit

Time-out period often relatively long long delay before

resending lost packet

Detect lost segments via duplicate ACKs Sender often sends

many segments back-to-back

If segment is lost there will likely be many duplicate ACKs

If sender receives 3 ACKs for the same data it supposes that segment after ACKed data was lost fast retransmit resend

segment before timer expires

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 64: School of Computing Science  Simon Fraser University

Transport Layer 3-64

event ACK received with ACK field value of y if (y gt SendBase) SendBase = y if (there are currently not-yet-acknowledged segments) start timer else increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) resend segment with sequence number y

Fast retransmit algorithm

a duplicate ACK for already ACKed segment

fast retransmit

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 65: School of Computing Science  Simon Fraser University

Transport Layer 3-65

TCP Flow Control

receive side of TCP connection has a receive buffer

speed-matching service matching the send rate to the receiving apprsquos drain rate app process may be

slow at reading from buffer

sender wonrsquot overflow

receiverrsquos buffer bytransmitting too

much too fast

flow control

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 66: School of Computing Science  Simon Fraser University

Transport Layer 3-66

TCP Flow control how it works

(Suppose TCP receiver discards out-of-order segments)

spare room in buffer= RcvWindow

= RcvBuffer-[LastByteRcvd - LastByteRead]

Rcvr advertises spare room by including value of RcvWindow in segments

Sender limits unACKed data to RcvWindow guarantees receive

buffer doesnrsquot overflow

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 67: School of Computing Science  Simon Fraser University

Transport Layer 3-67

TCP Connection Management opening

Application client Socket clientSocket = new

Socket(hostnameport number)

server when contacted by client Socket connectionSocket = welcomeSocketaccept()

TCP 3-way handshakeStep 1 client host sends TCP SYN

segment to server specifies initial seq no data

Step 2 server host receives SYN replies with SYNACK segment server allocates buffers specifies server initial seq

Step 3 client receives SYNACK replies with ACK segment which may contain data

client

SYN=1 seq= client_isn

server

SYN=1 seq=server_isn

ack=client_isn+1

SYN=0 seq=client_isn+1

ack=server_isn+1

conn request

conn granted

Q How would a hacker exploit TCP 3-way handshake to bring a server down

A SYN Flood DoS attack

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 68: School of Computing Science  Simon Fraser University

Transport Layer 3-68

TCP Connection Management closing

Application client closes socket clientSocketclose()

TCPStep 1 client end system sends TCP

FIN segment to server

Step 2 server receives FIN replies with ACK Closes connection sends FIN

Step 3 client receives FIN replies with ACK

Enters ldquotimed waitrdquo ndash may need to re-send ACK to received FINs

Step 4 server receives ACK Connection closed

client

FIN

server

ACK

ACK

FIN

closing

closing

closedti

med w

ait

closed

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 69: School of Computing Science  Simon Fraser University

Transport Layer 3-69

TCP Connection Management

TCP clientlifecycle

TCP serverlifecycle

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 70: School of Computing Science  Simon Fraser University

Transport Layer 3-70

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Congestion control 37 TCP congestion

control

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 71: School of Computing Science  Simon Fraser University

Transport Layer 3-71

Congestion Control Congestion sources send too much data for

network to handle different from flow control which is e2e

Congestion results in hellip lost packets (buffer overflow at routers)

bull more work (retransmissions) for given ldquogoodputrdquo

long delays (queueing in router buffers)bull Premature (unneeded) retransmissions

Waste of upstream linksrsquo capacity bull Pkt traversed several links then dropped at congested

router

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 72: School of Computing Science  Simon Fraser University

Transport Layer 3-72

Approaches towards congestion control

End-end congestion control

no explicit feedback from network

congestion inferred from end-system observed loss delay

approach taken by TCP

Network-assisted congestion control

routers provide feedback to end systems single bit indicating

congestion (SNA DECbit TCPIP ECN ATM)

explicit rate sender should send at

Two broad approaches towards congestion control

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 73: School of Computing Science  Simon Fraser University

Transport Layer 3-73

Chapter 3 outline

31 Transport-layer services

32 Multiplexing and demultiplexing

33 Connectionless transport UDP

34 Principles of reliable data transfer

35 Connection-oriented transport TCP segment structure reliable data transfer flow control connection

management

36 Principles of congestion control

37 TCP congestion control

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 74: School of Computing Science  Simon Fraser University

Transport Layer 3-74

TCP congestion control Approach

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Approach probe for usable bandwidth in network increase transmission rate until loss occurs then decrease Additive increase multiplicative decrease (AIMD)

time

Rat

e (C

ongW

in)

Saw toothbehavior probing

for bandwidth

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 75: School of Computing Science  Simon Fraser University

Transport Layer 3-75

TCP Congestion Control Sender keeps a new variable Congestion Window

(CongWin) and limits unacked bytes to

LastByteSent - LastByteAcked min CongWin RcvWin

For our discussion assume RcvWin is large enough

Sending Rate as function of CongWind is roughly Ignore loss and transmission delay

Rate = CongWindRTT (bytessec)

So rate and CongWind are somewhat synonymous

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 76: School of Computing Science  Simon Fraser University

Transport Layer 3-76

TCP Congestion Control Congestion occurs at routers (inside the network)

Routers do not provide any feedback for TCP

How can TCP infer congestion From its symptoms timeout or duplicate acks Define loss event equiv timeout or 3 duplicate acks TCP decreases its CongWin (rate) after a loss event

TCP Congestion Control Algorithm three components AIMD additive increase multiplicative decrease slow start Reaction to timeout events

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 77: School of Computing Science  Simon Fraser University

Transport Layer 3-77

AIMD additive increase (congestion avoidance phase)

increase CongWin by 1 MSS every RTT until loss detected TCP increases CongWin by MSS x (MSSCongWin) for every

ACK received Ex MSS = 1460 bytes and CongWin = 14600 bytes With every ACK CongWin is increased by 146 bytes

multiplicative decrease cut CongWin in half after loss

8 Kbytes

16 Kbytes

24 Kbytes

time

congestionwindow

Con

gWin

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 78: School of Computing Science  Simon Fraser University

Transport Layer 3-78

TCP Slow Start When connection begins CongWin = 1 MSS

Example MSS = 500 bytes amp RTT = 200 msec initial rate = CongWinRTT = 20 kbps

available bandwidth may be gtgt MSSRTT desirable to quickly ramp up to respectable rate

Slow start When connection begins increase rate exponentially

fast until first loss event How can we do that double CongWin every RTT How Increment CongWin by 1 MSS for every ACK received

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 79: School of Computing Science  Simon Fraser University

Transport Layer 3-79

TCP Slow Start (contrsquod)

Increment CongWin by 1 MSS for every ACK

Summary initial rate is slow but ramps up exponentially fast

Host A

one segment

RTT

Host B

time

two segments

four segments

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 80: School of Computing Science  Simon Fraser University

Transport Layer 3-80

Reaction to a Loss event

TCP Tahoe (Old) Threshold = CongWin 2 Set CongWin = 1 Slow start till threshold Then Additive Increase congestion avoidance

TCP Reno (most current TCP implementations) If 3 dup acks fast retransmit

bull Threshold = CongWin 2bull Set CongWin = Threshold fast recovery bull Additive Increase

Else timeoutbull Same as TCP Tahoe

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 81: School of Computing Science  Simon Fraser University

Transport Layer 3-81

Reaction to a Loss event (contrsquod)

Why differentiate between 3 dup acks and timeout

3 dup ACKs indicates network capable of delivering some segments

timeout indicates a ldquomore alarmingrdquo congestion scenario

3 dup acks

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 82: School of Computing Science  Simon Fraser University

Transport Layer 3-82

TCP Congestion Control Summary Initially

Threshold is set to large value (65 Kbytes) has not effect

CongWin = 1 MSS

Slow Start (SS) CongWin grows exponentially

till a loss event occurs (timeout or 3 dup ack) or reaches Threshold

Congestion Avoidance (CA) CongWin grows linearly

3 duplicate ACK occurs

Threshold = CongWin2 CongWin = Threshold CA

Timeout occurs

Threshold = CongWin2 CongWin = 1 MSS SS till Threshold

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 83: School of Computing Science  Simon Fraser University

Transport Layer 3-83

TCP throughput

Whatrsquos the average throughout of TCP as a function of window size and RTT Ignore slow start

Let W be the window size when loss occurs

When window is W throughput is WRTT Just after loss window drops to W2

throughput to W2RTT

Average throughout 075 WRTT

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 84: School of Computing Science  Simon Fraser University

Transport Layer 3-84

TCP Futures Throughput in terms of loss rate (homework)

Example 1500 byte segments 100ms RTT want 10 Gbps throughput

Requires loss rate to be L = 210-10 (wow)

New versions of TCP for high-speed needed

LRTT

MSS221

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 85: School of Computing Science  Simon Fraser University

Transport Layer 3-85

Fairness goal if K TCP sessions share same bottleneck link of bandwidth R each should have average rate of RK

TCP connection 1

bottleneckrouter

capacity R

TCP connection 2

TCP Fairness

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 86: School of Computing Science  Simon Fraser University

Transport Layer 3-86

Why is TCP fair

Two competing sessions Additive increase gives slope of 1 as throughout increases multiplicative decrease decreases throughput proportionally

R

R

equal bandwidth share

Connection 1 throughputConnect

ion 2

th

roughput

congestion avoidance additive increaseloss decrease window by factor of 2

congestion avoidance additive increaseloss decrease window by factor of 2

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 87: School of Computing Science  Simon Fraser University

Transport Layer 3-87

Fairness (more)

Fairness and UDP Multimedia apps

often do not use TCP do not want rate

throttled by congestion control

Instead use UDP pump audiovideo at

constant rate tolerate packet loss

Research area TCP friendly

Fairness and parallel TCP connections

nothing prevents app from opening parallel connections between 2 hosts

Web browsers do this Example link of rate R

supporting 9 cnctions new app asks for 1 TCP

gets rate R10 new app asks for 11 TCPs

gets R2

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary
Page 88: School of Computing Science  Simon Fraser University

Transport Layer 3-88

Chapter 3 Summary principles behind transport

layer services multiplexing

demultiplexing reliable data transfer flow control congestion control

instantiation and implementation in the Internet UDP TCP

Next leaving the network

ldquoedgerdquo (application transport layers)

into the network ldquocorerdquo

  • School of Computing Science Simon Fraser University
  • Chapter 3 Transport Layer
  • Chapter 3 outline
  • Transport services and protocols
  • Transport vs network layer
  • Internet transport-layer protocols
  • Slide 7
  • Multiplexingdemultiplexing
  • How demultiplexing works
  • Connectionless demux
  • Connection-oriented demux
  • Connection-oriented demux (cont)
  • Connection-oriented demux Threaded Web Server
  • Slide 14
  • UDP User Datagram Protocol [RFC 768]
  • UDP more
  • UDP checksum
  • Internet Checksum Example
  • Slide 19
  • Principles of Reliable data transfer
  • Slide 21
  • Slide 22
  • Reliable data transfer getting started
  • Slide 24
  • Rdt10 reliable transfer over a reliable channel
  • rdt20 channel with bit errors
  • rdt20
  • rdt20 has a fatal flaw
  • rdt21 sender handles garbled ACKNAKs
  • rdt21 receiver handles garbled ACKNAKs
  • rdt21 discussion
  • rdt22 a NAK-free protocol
  • rdt30 channels with errors and loss
  • rdt30 sender
  • rdt30 in action
  • Slide 36
  • Performance of rdt30
  • Performance of rdt30 example
  • Pipelined protocols
  • Pipelining increased utilization
  • Go-Back-N
  • GBN sender extended FSM
  • GBN receiver extended FSM
  • GBN in action
  • Slide 45
  • Selective Repeat
  • Selective repeat sender receiver windows
  • Selective repeat
  • Selective repeat in action
  • Slide 50
  • TCP Overview RFCs 793 1122 1323 2018 2581
  • TCP segment structure
  • TCP seq rsquos and ACKs
  • TCP reliable data transfer
  • TCP sender events
  • TCP sender (simplified)
  • TCP retransmission scenarios
  • TCP retransmission scenarios (more)
  • TCP Round Trip Time and Timeout
  • Slide 60
  • Example RTT estimation
  • Slide 62
  • Fast Retransmit
  • Fast retransmit algorithm
  • TCP Flow Control
  • TCP Flow control how it works
  • TCP Connection Management opening
  • TCP Connection Management closing
  • TCP Connection Management
  • Slide 70
  • Congestion Control
  • Approaches towards congestion control
  • Slide 73
  • TCP congestion control Approach
  • TCP Congestion Control
  • Slide 76
  • AIMD
  • TCP Slow Start
  • TCP Slow Start (contrsquod)
  • Reaction to a Loss event
  • Reaction to a Loss event (contrsquod)
  • TCP Congestion Control Summary
  • TCP throughput
  • TCP Futures
  • TCP Fairness
  • Why is TCP fair
  • Fairness (more)
  • Chapter 3 Summary