Uwe Glässer Software Technology Lab School of Computing Science Simon Fraser University
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Transcript of 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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-
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
-