Introduction to VoIP Technology Tutorials

Copyright © 2004 OPNET Technologies, Inc. Confidential, not for distribution to third parties.

Introduction to VoIPTechnology Tutorials

Session 1819

Copyright © 2004 OPNET Technologies, Inc` Confidential, not for distribution to third parties. 2

1819 Introduction to VoIP

What is VoIP?

Why is it used?

How is it used? Applications and architectures

How does VoIP work? Protocols

What do VoIP calls sound like? QoS

How can I make sure that VoIP deployments will work properly? Modeling and simulation

Agenda



What is VoIP?

Carrying voice conservations over Internet protocol packet networks Private

Public

There are other flavors of packetized voice: VoATM

VoFR

IP Network



Why Use VoIP?

Cost savings

Integrated data and voice networks

Device interoperability using standards-based protocols

Flexibility in deriving new services



Traditional Voice Versus VoIP

A traditional T1 can carry 24 telephone calls simultaneously

With VoIP, a T1 can carry 64 calls simultaneously!

G.729 8kbps compression, 20 msec frame size = 24 kbps

1544 / 24 = 64 calls per T1

T1 = 1544 kbps, DS0 = 64 kbps,

1544 / 64 = 24 DS0 per T1



Devices

IP Telephones A telephone that directly connects to an IP network

Gateways Provide bulk conversion of connections between

signaling domains: PSTN connections to VoIP connections One VoIP signaling domain to another

Servers Handle registration, authentication, telephone

number to IP address conversion, bandwidth management, etc.



How is VoIP Used?

Applications and architectures

Consumer

Campus

Enterprise

Service Provider



Consumer: IP-to-IP

Uses PC software to make calls over public and private internets

Free!!But, no quality of service guaranteesExamples:

Microsoft NetMeetingTM

SkypeTM

Hybrids PC2PhoneTM

The InternetPC

PCModemModem



Campus Applications – IP PBX

Connecting office telephones to PBX with VoIP linksVendors

Cisco Nortel

PSTN

LAN Switches

IP Telephones

Call Manager &IP-to-PSTN Gateway



Enterprise Applications – Toll Bypass

Connecting enterprise PBXs with VoIP links to avoid paying for long distance charges Vendors:

Nortel NEC Avaya Toshiba Ericsson Cisco

Private Data Network or VPN

Public Switched Telephone Network

(PSTN)

PBX PBXRouter Router



Service Provider Applications – Local AccessUsing broadband access to provide local and long distance

telephone serviceExample Services:

Vonage ATT CallVantageTM

Packet8 Broadvox Time Warner Cable

PC

BroadbandModem

Splitter

Ordinary Telephone

Broadband Service

ProviderPSTN

Access Provider

ISP



Service Provider Applications – Trunking

Carrying voice traffic between switches over long haul networkAllows for consolidation with data networksExample Hardware:

Nortel Sonus

*LATA = local access and transport area

LATA #2

Private Data Network

LATA #1

LATA #3

LATA #4

LATA #5

LATA #6

PSTN

Gateway Gateway



What is VoIP?

Why is it used?





Agenda



IP Phone IP Phone

How Does VoIP Work?

Gatekeeper

LAN

1. Caller dials 555-1234 3. Gatekeeper responds with the IP address of the called party

4. Caller sends a call setup message to the called party

5. Called party accepts the call by picking up the telephone receiver. An “accept” message is sent back to the caller.

6. Voice packets flow between IP telephones

2. Gatekeeper performs authentication, call admission control, and address translation



IP Phone

How Does VoIP Work? (IP-to-PSTN)

Gatekeeper

LAN

1. Caller dials 555-1234

3. Gatekeeper responds with the IP address of the gateway

4. Caller sends a call setup message to the gateway


PSTNOrdinary Telephone

IP-to-PSTNGateway

7.Gateway converts the PSTN accept message into VoIP accept message and sends it back to the caller.

5. Gateway converts the VoIP signaling message to PSTN signaling message

6. Called party accepts the call by picking up the telephone receiver. An “accept” message is sent back to the gateway.

8. Voice packets flow between IP telephone and gateway. Gateway converts between packet data and timeslot data.



Protocol Soup

Signaling H.323, SIP, MGCP, H.248, SCCP, etc.

CODECs G.711, G.723, G.729

Transport RTP, RTCP, CRTP, ECRTP, UDP, IP

Other RSVP, DiffServ, IntServ, MPLS, DNS, COPS (policy), Radius &

Diameter (authentication)



Signaling Protocols

H.323 Distributed architecture Used for video conferencing, but also VoIP

SIP – Session Initialization Protocol Distributed architecture IETF RFC 2543

MGCP – Media Gateway Control Protocol Centralized architecture IETF RFC 2705

H.248 Centralized architecture Extends MGCP Collaboration between ITU and IETF Also known as RFC 2885, Megaco

SCCP – Skinny Client Control Protocol Cisco proprietary For use with Cisco CallManager

H.323

SIP

Distributed

MGCP

H.248

Centralized



H.323 Details

ITU umbrella standard for packet-based multimedia communication systems Audio CODECs Video CODECs H.255 registration, admission, and status (RAS) H.225 call signaling H.245 control signaling Real-time transport protocol (RTP) Real-time control protocol (RTCP)

Early standard

Complex

Transport Protocols

RTP

RTCP H.255

RAS

H.225

Call Signaling

H.245

Control Signaling

T.120

Data

G.711

G.729

G.723

H.261

H.263

Call Manager ApplicationAudio Video



H.323 Components

Terminals Hardware or software running H.323

protocols

Gateway Connects different networks

H.323-to-PSTN H.323-to-{other VoIP signaling protocol}

Gatekeeper (optional) Address translation Admission control Bandwidth control Zone control



H.323 Call Setup

Gatekeeper

LAN

1. Caller dials 555-1234 3. Gatekeeper responds with the IP address of the called party

4. Caller sends a call setup message to the called party

6. Voice packets flow between IP telephones


IP Phone IP Phone (555-1234)

5. Called party accepts the call by picking up the telephone receiver. An “accept” message is sent back to the caller.

H.323 Messages



Session Initiation Protocol (SIP) Details

Recent standard

Simpler then H.323

Also used for video conferencing, network gaming, instant messaging

Similar to HTTP, textual coding

Uses URLs for addressing: sip:[email protected]

sip:[email protected]?subject=callme

sip:[email protected]

tel:+1-919-555-1234

DTMFs carried in signaling message



SIP Call Setup

SIP Proxy

IP Network

INVITE sip:[email protected]: [email protected]:[email protected]:[email protected]

INVITE sip:[email protected]: [email protected]:[email protected]:[email protected]

ACK [email protected] packets flow between IP telephones

Proxy for sip.com gets location information for called party.

IP Phone ([email protected])

IP Phone ([email protected])

OK 200From: [email protected]:[email protected]:[email protected]

OK 200From: [email protected]:[email protected]:[email protected]



MGCP/H.248/Megaco Details

Based on master/slave principal More palatable to telco's Easier to rollout new feature since only the servers need to be updated,

not the individual telephones

Media Gateway Controller

Signaling Gateway

PSTN

H.248 Messages

Call Control (SIP, H.323, etc.)

SS7, etc.

Trunks

Media Gateway Controller

IP PhoneMedia Gateway



Break!!!

Break!



CODECS

Voice codecs create blocks of data at fixed intervals Usually 10 ms

Each block contains a fixed number of bytes depending on the coding scheme used 10-80 bytes/block

Codecs can typically be parameterized to put a given number of voice data bytes into a single IP packet 10, 20, 30, …, 240 bytes

Bandwidth saving techniques Silence suppression Compression

Tradeoffs Small packets = less delay, but more layer 2/3 overhead Large packets = more delay, less layer 2/3 overhead



Typical CODEC Behavior



CODEC Characteristics

Codec Compression Method

Codec Bit Rate

Block Length

Block Size

(bytes)

Blocks per

Packet

Voice Call Bandwidth Required

(Excl. L2 o/h)

Mean Opinion Score

Compression Delay (ms)

g711alaw PCM 64000 10 ms 80 2 80000 4.1 0.75

g711ulaw PCM 64000 10 ms 80 2 80000

g723ar53 ACELP 5300 10 ms 7 2 22000 3.65 30

g723ar63 MP-MLQ 6300 10 ms 8 2 23000 3.9 30

g723r53 ACELP 5300 10 ms 7 2 22000

g723r63 MP-MLQ 6300 10 ms 8 2 23000

g726r16 ADPCM 16000 10 ms 20 2 32000

g726r24 ADPCM 24000 10 ms 30 2 40000

g726r32 ADPCM 32000 10 ms 40 2 48000 3.85 1

g728 LD-CELP 16000 10 ms 20 2 32000 3-5

g729r8 CS-ACELP 8000 10 ms 10 2 24000 3.92 10

g729br8 CS-ACELP 8000 10 ms 10 2 24000 3.7 10



Real-time Transport Protocol (RTP)

Media content typeTalk spurtsSender identificationSynchronizationLoss detectionSegmentation and reassemblySecurity (encryption)

V P X PayloadM Sequence Number

Timestamp

Synchronization Source Identifier (SSRC)

Payload



RTP Control Protocol (RTCP)

Used for monitoring the quality of a session

Transferring that information to all of the participants in the session

Provides minimal session control

Sent on different port number from RTP

Messages: Sender Reports: Information about sent data, synchronization timestamp

Receiver Reports: Information about received data, losses, jitter and delay

Source Description:Name, Email, Phone, Identification

Bye: Explicit leave indication

Application defined parts: Parts for experimental functions



Compressed RTP

Technique for reducing the bandwidth requirements for RTP-UDP-IP headers

Reduces all three headers from 40 bytes to 2-4 bytes

RTP Header = 12 bytes

UDP Header = 8 bytes

IP Header = 20 bytes

Utilizes the fact that much the headers’ contents remain the same from packet

to packet

Critical for low-speed uplinks

Versions: RFC 2508, CRTP for low-speed serial links

RFC 3545, Enhanced CRTP for high delay, packet loss, and reordering



Other Issues

Interoperability between signaling protocols Gateways can convert between protocols

Handling modem and fax traffic Detection needed at gateway T.37/T.38 Fax Delivery of IP Modems must use G.711 with no echo

cancellation and no high pass filter

VoIP Network

IP-to-PSTNGateway

IP-to-PSTNGateway

Fax

Modem

Fax

Modem

PSTN PSTN



What is VoIP?

Why is it used?





Agenda



What Do VoIP Calls Sound Like?

Sound quality depends on many factors Telephone quality

Type of CODEC used

Higher compression leads to lower quality

Network performance

Quality of Service Metrics



Subjective Versus Objective Quality ScoringMean Opinion Score (MOS)

A telephone industry standard for measuring voice quality Based on users’ perceptions of voice quality

Excellent = 5, Good = 4, Fair = 3, Poor = 2, Bad = 1 MOS should be > 4.0

E-model, ITU G.107 Predicts the MOS based on

CODEC characteristics Packet loss Delay Jitter

0

0.5

1

1.5

2

2.5

3

3.5

4

4.5

5

CODEC #1 CODEC #2

Excellent

Good

Poor

Bad

Fair



Quality of Service Metrics

Packet Loss – What percentage of the packets are dropped Should be less than 1%

Delay – How much time elapses between when an utterance is spoken and when it is played back at the receiver Must be less than 150 ms for real-time conversations

Jitter – The variability in the delay Must be less than 30 ms De-jitter buffer helps fix the problem, but adds to the overall delay



Example of Delay Budget

Delays of less than 150 ms are sought But the fixed components of delay can be high Careful control of the variable components (queuing) required

Delay Component Fixed/Variable Delay (msec)

Codec-Related

g729a Compression Delay fixed 5

g729a Sampling Delay (10 ms x 2) fixed 20

Queuing Delay on Trunk variable 5

Transmission Delay fixed 3

Propagation Delay fixed 25

Queuing at Intermediate Hops variable 20

De-jitter buffer fixed 50

Total of Fixed Delays 103

Total of Variable (Queuing) Delays 25

Total Delay 128



Active Quality Measurement Systems

Use active network to monitor the QoS of a VoIP Network

Examine actual calls check performance

Set up extra calls on real network to test performance

Monitoring software is embedded in gateways and other devices

Use E-model to estimate MOS

Vendors Psytechnics RADCOM Agilent



QoS Mechanisms – Queuing

Queuing – Mechanisms for giving different treatment to different types of packets First In, First Out (FIFO)

Default behavior

Priority queuing (PQ)

Strict ordering of queues

Weighted Fair Queuing (WFQ)

Each queue gets a percentage of the bandwidth during congestion

Combination

A single high-priority queue + WFQ + best-effort queue



FIFO Queue Example

Voice packets can get delayed or even dropped due to interaction with data flows

Voice Flow

Data Flows

FIFO Queue

Packets lost due to tail drop during congestion

As the queue length grows, so does the average delay

The varying length of the queue adds to the jitter

Mu

ltiplexer



Example of WFQ + Priority Queue

Voice packets are always transmitted first via the “Priority FIFO Queue”

Voice Flow

Data Flows

Priority FIFO Queue

Classifier

WFQ Queues Sch

edu

ler

Best-effort Queue

…



QoS Mechanisms

Ethernet QoS – 802.1pIntServ – A mechanism for a reserving resources on

devices via RSVP signaling Fine-grained Not scalable

DiffServ – A static mechanism for marking packets at the edge of the network and giving per-class treatment within the network Coarse Scalable No signaling

MPLS-DiffServ-TE Using label switched paths to control the paths that packets take

through the network as well as the treatment they receive at each hopCall Admission Control (CAC)

Gatekeeper/Proxy function for limiting number of calls in system



What is VoIP?

Why is it used?





Agenda



Deployment Considerations

QoS strategy

Server location Signaling latency issues

Load balancing

Redundancy

Dial plan

PSTN backup

Electrical power

Connectivity to Voice Mail and other Integration Voice Response (IVR) systems

Cooperation between telecom and data teams



Modeling and Simulation

Configuration analysis Process configuration files for errors and

security problems Readiness assessment

Propagation delay prediction Failure analysis

Capacity planning Using flow analysis to determine the appropriate

link sizes in a VoIP network Voice traffic conversion: erlangs to bits/sec

QoS configuration planning Setting queue sizes

Voice quality analysis Using discrete event simulation (DES) to model

packet loss, delay, jitter of voice calls Protocol modeling

Using ACE or DES to model and verify VoIP signaling protocols and signaling latencies



Useful VoIP Links

Transition to VoIP in campus http://www.cisco.com/warp/public/cc/so/neso/vvda/avvid/ttpnp_bc.pdf

Market research http://www.sonusnetworks.com/contents/brochures/solutions/Market_Impact.pdf

General information http://www.voip-news.com/

http://www.voip-info.org/

SIP information http://www.cs.columbia.edu/sip/

CODEC calculator http://www.voip-calculator.com/calculator/lipb/



Documentation References

H.323 ITU Standard for Voice/Video over IP

SIP – Session Initialization Protocol, IETF RFC 2543

MGCP – Media Gateway Control Protocol, IETF RFC 2705

H.248, Megaco, IETF RFC 2885

SCCP – Skinny Client Control Protocol

RTP – Real-time Transport Protocol, IETF RFC 1889

RTCP – RTP Control Protocol, IETF RFC 1889

CRTP for low-speed serial links, RFC 2508

Enhanced CRTP for high delay, packet loss, and reordering, RFC 2508

ITU-T.37 – Procedures for the Transfer of Facsimile Data Via Store-and-forward on the Internet

ITU-T.38 – Procedures for Real-time Group 3 Facsimile Communication over IP Networks



Related OPNETWORK Sessions

1346 Planning and Analyzing VoIP Deployments Thursday, 14:00-16:00, Atrium Ballroom B

1352 Case Studies: VoIP and Circuit-to-Packet Thursday, 14:00-16:00, Continental A

1806 Introduction to QoS Mechanisms Thursday, 14:00-16:00, Continental C

1337 Case Studies: QoS I

1338 Case Studies: QoS II Thursday, 16:00-18:00, Polaris C



Take-Away Points

VoIP can take many forms

Toll-bypass, PBX, access, trunking

Many signaling protocols and architectures will be deployed

Providing QoS guarantees is critical to VoIP success

Modeling and simulation tools can help address these issues

The VoIP market is growing – Get prepared!

Introduction to VoIP Technology Tutorials

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