understanding voice over ip protocols

1© 2002, Cisco Systems, Inc. All rights reserved.4426_02_2002_c1

Understanding Voice over IP Protocols

Cisco Systems—Service Provider Solutions Engineering

February, 2002


Topics to Discuss

• History of VoIP

• VoIP—Early Adopters

• VoIP—Standards and Standards Bodies

• VoIP—Making Sense of the Protocols

• “The Great Voice Myth”

• VoIP—Protocol Challenges

• Summary


Why Move to VoIP?

• Cost savings—toll bypass

• Open standards—H.323, SIP, MGCP

• Multi-vendor interoperability

• Integrated IP voice and data networks


Cisco Packet Voice Architecture

TDM/Circuit Switch

TDM/Circuit Switch

Digital Trunk Digital Trunk SubsystemSubsystem

Line Line ConcentrationConcentration

AdministrationAdministrationMaintenanceMaintenance

BillingBilling

Call ControlCall ControlConnection ControlConnection Control

FeaturesFeatures

Common ChannelCommon ChannelSignaling ComplexSignaling Complex

Sw

itch

ing

Net

wo

rk

StandardsStandards--Based Based Packet Infrastructure LayerPacket Infrastructure Layer

(IP, ATM)(IP, ATM)

Open Call Control LayerOpen Call Control Layer(SIP, H.323, MGCP, etc.)(SIP, H.323, MGCP, etc.)

Open Service Open Service Application LayerApplication Layer

(JAIN, AIN, TAPI,(JAIN, AIN, TAPI,JTAPI, XML etc.)JTAPI, XML etc.)

Open/Standard Interface

Open/Standard Interface


Topics to Discuss

• History of VoIP






• Summary


Early Adopters—Advanced Services and Toll-Bypass

• Regulatory opportunities allowed for toll-bypass

• PC-to-phone, calling-card and international fax services

• Cisco-based carriers used standard protocols, but not all carriers implemented standards

• Inter-carrier connections had protocol interoperability challenges


Topics to Discuss

• History of VoIP






• Summary


Making the Rules for VoIP

• IETF (Internet Engineering Task Force)The community of engineers that standardizes the protocols that define how the Internet and Internet Protocols work. http://www.ietf.org/

• ITU (International Telecommunications Union)An international organization within the United Nations System where governments and the private sector coordinate global telecom networks and services. http://www.itu.int/home/index.html


Defining the VoIP Protocols

• H.323An ITU Recommendation that defines “Packet-based multimedia communications systems”. H.323 defines a distributed architecture for creating multimedia applications, including VoIP

• SIPDefined as IETF RFC 2543. SIP defines a distributed architecture for creating multimedia applications, including VoIP

• MGCPDefined as IETF RFC 2705. MGCP defines a centralized architecture for creating multimedia applications, including VoIP

• H.248An ITU Recommendation that defines “Gateway Control Protocol”. H.248 is the result of a joint-collaborate with the IETF. H.248 defines a centralized architecture, and is also known as “Megaco”

• MegacoDefined as IETF RFC 2885. Megaco defines a centralized architecture


Topics to Discuss

• History of VoIP






• Summary


H.324Terminal

H.324Terminal

H.323Gatekeeper

H.323Gatekeeper

Packet Network

H.323Terminal

H.323Terminal

H.323Gateway

H.323Gateway

H.323MCUH.323MCU Scope of

H.323

PSTN ISDN

V.70Terminal

V.70Terminal

SpeechTerminalSpeechTerminal

H.320Terminal

H.320Terminal

SpeechTerminalSpeechTerminal

H.323 Components

e

GK


Scope of H.323 Recommendation

Video CodecH.261, H.263

Call ControlH.225.0

System ControlSystem Control

Audio CodecG.711, G.722,G.723, G.728,

G.729

RAS ControlH.225.0

RTP

RTCP

H.245 Control

Audio I/OEquipment

User Data Applications

T.120, etc.

SystemControl

UserInterface

Video I/OEquipment

H.225Layer

UDP

TCP

UDP

IP

ReceivePain

Delay(Sync)


Media (UDP)RTP StreamRTP StreamRTCP StreamRTCP Stream

H.323Endpoint A

H.245 (TCPDynamic Port)Open Logical Channel

H.225 (TCP Port 1720)Setup

Alerting / Connect

Open Logical Channel Acknowledge

Capabilities Exchange / MSD

RTP StreamRTP Stream

H.323Endpoint B

VVVV

H.323 Signaling


Gatekeeper A Gatekeeper B

RRQ/RCF

ARQ

RRQ/RCF

LRQ

IP Network

Phone A

Gateway A Gateway B

H.225 (Q.931) Setup

H.225 (Q.931) Alert and Connect

H.245

RTP

ACF

LCF

VV

Basic H.323 Call

VV

ARQ

ACF

Phone B


MidwestZoneMidwestZone

ChicagoGWChicagoGW

LocalPSTNLocalPSTN

Deploying H.323 Networks

WestZoneWestZone

• Addition of new rate centers

• Minimizes GK configuration• Addition of new zones

DGK• Addition of new NPAs

LAGW #1

LAGW #1

RateCenter #1

RateCenter #1

RateCenter #1

RateCenter #1

LAGW #2

LAGW #2

Intra-LATAToll

Intra-LATAToll

GK GK GK

EastZoneEastZone

NYGWNYGW

LocalPSTNLocalPSTN


MGCP/H.248/Megaco—Architectures

PSTN

Call Agent

SS7

AccessGateway

MGCP / H.248 / Megaco

IMT

PRI

RTP

Call Agent

PSTN

PSTN


Deploying MGCP/H.248/Megaco Networks

Modem Dial-up TrafficTraditional TDM Traffic

CA

MGCPand ISDNBackhaul

Service Provider's

TDMNetwork

Service Provider's

PacketNetwork

MGMG

TDMVoice

NAS/VoIP

ISDN/PRI

OSSOSS

Billing and MeasurementServer

Billing and MeasurementServer

SLT

VoIP

MG

SS7Backhaul

SS7

STP

PSTNIMTs


SIP Architecture

Legacy PBX

SIP User Agents (UA)

Application Services

eMail LDAP Oracle XML

SIPSIP

RTP(Media)

SIP

CPLCPL3pcc

PSTN

CAS or PRI

INTELLIGENT

SERVICES

SIP Proxy, Registrar & Redirect Servers


SIP VoIP Network

Calling Party

PSTN

Called Party

PSTN

SignalingSignaling

Bearer Or Media

Bearer Or Media

SIP Signaling

Media (UDP)

INVITE

100 Trying

180 Ringing

100 Trying

RTCP StreamRTCP Stream

INVITE

200 OK200 OK

180 Ringing

ACK ACK

SIP Signaling and SDP Signaling(UDP or TCP)


SIP Servers/ServicesSIP Servers/Services

SIP User Agents

Registrar RedirectLocationDatabase

SIP Proxy

SIP Servers/Services

REGISTER“Here I am”

INVITE“I want to talk to another UA

Proxied INVITE“I’ll handle it for

you”

“Where is this name/phone#?”

3xx Redirection“They moved,

try this address”

SIP User Agents SIP-GW


Central ZoneCentral Zone

ChicagoPOPChicagoPOP

West ZoneWest Zone

SFPOPSFPOP

PSTN312

East ZoneEast Zone

NYPOPNYPOP

PSTN212

PSTN415

Deploying SIP Networks

IP Network


Topics to Discuss

• History of VoIP






• Summary


Voice Myths

• Networks can only be built one way

• Networks will only use one protocol

• All networks will converge

• VoIP allows centralized or distributed architectures

• H.323, SIP, MGCP and H.248/Megaco will all be present in VoIP networks

• Networks will converge to IP

Myths Facts


Topics to Discuss

• History of VoIP






• Summary


Interconnecting VoIP Networks

??SIPSIPH.323

H.323

MGCPMGCPH.248H.248

MegacoMegaco


Connecting VoIP to SS7/C7 Networks

INVITEACKSDPSDP

H.323

MGCPMGCP

SIPSIP

IAM

CRCXACKSDPSDP

H.225 Setup (ANI,DN)Proceeding

H.245H.245


VoIP Interworking Issues

• Service interworkingE.g.: H.450 <-> SIP <-> MGCP

• Media interworkingEnd-to-end codec negotiation

• Bearer interworkingEnd-to-end fax, modem, DTMF


VoIP Interworking

• Bearer levelModem (relay/passthru)

Fax (relay/passthru)

T.38

T.37

DTMF (relay/passthru)

• Media levelCodec (negotiation, selection)

• Service translation issues

Call deflection

Park/hold

• Signal issuesSDP

H.245


Fax and Modem Passthru Mechanisms

• Modem and fax are control mechanisms based on PLL (Phase Locked Loops)

• They are both time sensitive

• Highly sensitive to packet network impairments: Jitter

Packet loss

Delay

• Susceptible to clock slew (clock sync differences between gateways)


IPCloud

Voice Gateway

Voice Gateway

G.711µ

G.729

G.729

G.711µ

PCMG.711µ

PCMG.711µ

Passthru Simplified

DSPDSP

DSPDSP


What Is Modem Passthru?

• It is the transport of modem signals (modulation, error correction and compression) through a packet network using PCM encoded packets


Modem Passthru (Cont.)

• Modem tone detection (<= V.90)

• Switchover signaling

• No VAD

• EC off

• RTP payload redundancy (10ms packetization) RFC2198 (optional)


Modem Passthru Issues

• Consecutive packet drops (loss) cause retrain

• Consecutive drops during retrain causes disconnect

• Variation of delay (jitter) has quite an effect

• Jitter (at 10%) is a conservative estimate—Since jitter mostly impacts performance with packet loss


What Is Modem Relay?

• Modem relay involves demodulating the modem signal at ingress gateway

• Passing this data as packet data to terminating gateway

• Re-modulating the data and passes it to the receiving modem


Fax Relay—T.38

• Real-time

• Also called demod/remod

• Can be used in H.323/MGCP/SIP signaling

• Delivers fax data over UDP streams (uses same RTP port)—reuses voice UDP ports

• Fallback to proprietary mode

• Method of encoding the T.30 and T.4 into packets

PSTNPSTNPSTNPSTN

IP

T.30 T.30UDP


DTMF

• What is DTMF

• Why is it required? and where is it used?

• How do you transport it in IP?

• DTMF implementation


DTMF (Cont.)

• In TDM world, all voice traffic is sent as uncompressed 64Kbs PCM streams; anything sent on that circuit is an untouched stream of bits; (e.g., voice speech, modem tones, fax tones, and DTMF digits)

• DSP codecs designed to interpret human speech, can distort DTMF tones (machine-tones)

• High b/w codecs less likely to distort

• Distortion causes problems with voicemail and IVR systems


DTMF Schemes with VoIP Protocols

H.323H.323MGCP, H.248,

Megaco

MGCP, H.248,

MegacoSIPSIP

In-BandIn-Band In-BandIn-Band In-BandIn-Band In-BandIn-Band

Out-of-BandOut-of-Band

Cisco RTP, H.245 Alphanum,H.245 Signal, AVT Tones RFC2833

Cisco RTP, H.245 Alphanum,H.245 Signal, AVT Tones RFC2833

Cisco RTP,NSE, NTE,RFC2833

Cisco RTP,NSE, NTE,RFC2833 RFC2833RFC2833


Topics to Discuss

• History of VoIP






• Summary


Summary

• Understand the possibilities and the issues

• Avoid protocol/product based bias

• Decide on application

• Consider market and business drivers

• Deploy what’s possible today

• Choose signaling protocol depending on services intended to be offered

• Many possibilities—stay tuned


Crystal Ball on VoIP

• All three protocols (or its variations) are here for the long run

• Changes/enhancements will be made

• IP will be the core


Reference URLs

• ITU: www.itu.org

• IETF: www.ietf.org

• SIP: www.cs.columbia.edu/~hgs/sip/

• H.323: www.packetizer.com/iptel/h323/

• MGCP:www.softswitch.org/asp/techlibrary_protocol.asp?page=techlibrary

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