Understanding Voice over IP Protocols

Cisco SystemsService Provider Solutions Engineering February, 2002

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Topics to Discuss
History of VoIP VoIPEarly Adopters VoIPStandards and Standards Bodies VoIPMaking Sense of the Protocols The Great Voice Myth VoIPProtocol Challenges Summary

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Why Move to VoIP?

Cost savingstoll bypass Open standardsH.323, SIP, MGCP Multi-vendor interoperability Integrated IP voice and data networks

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Cisco Packet Voice Architecture

Open Service Application Layer TDM/ Circuit Switch
Switching Network
Line Concentration Call Control Connection Control Features

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

Open/Standard Interface
Open Call Control Layer
(SIP, H.323, MGCP, etc.)

Digital Trunk Subsystem

Common Channel Signaling Complex

Administration Maintenance Billing

Open/Standard Interface
Standards-Based StandardsPacket Infrastructure Layer
(IP, ATM)

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Topics to Discuss
History of VoIP VoIPEarly Adopters VoIPStandards and Standards Bodies VoIPMaking Sense of the Protocols The Great Voice Myth VoIPProtocol Challenges Summary

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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

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Topics to Discuss
History of VoIP VoIPEarly Adopters VoIPStandards and Standards Bodies VoIPMaking Sense of the Protocols The Great Voice Myth VoIPProtocol Challenges Summary

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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

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Defining the VoIP Protocols

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

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

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

H.248
An 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

Megaco
Defined as IETF RFC 2885. Megaco defines a centralized architecture
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Topics to Discuss
History of VoIP VoIPEarly Adopters VoIPStandards and Standards Bodies VoIPMaking Sense of the Protocols The Great Voice Myth VoIPProtocol Challenges Summary

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H.323 Components
GK

H.323 MCU

Scope of H.323
e

H.323 Gatekeeper

Packet Network

H.323 Terminal

H.323 Gateway

PSTN

ISDN

V.70 Terminal

H.324 Terminal

Speech Terminal

H.320 Terminal

Speech Terminal

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Scope of H.323 Recommendation

Video I/O Equipment
Video Codec H.261, H.263

Audio I/O Equipment User Data Applications T.120, etc.

Audio Codec G.711, G.722, G.723, G.728, G.729

Receive Pain Delay (Sync)

RTP UDP RTCP

H.225 Layer
System Control H.245 Control

IP
TCP

System Control User Interface

Call Control H.225.0 RAS Control H.225.0

UDP

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H.323 Signaling

H.323 Endpoint A

H.323 Endpoint B
Setup Alerting / Connect

H.225 (TCP Port 1720)

Capabilities Exchange / MSD Open Logical Channel Open Logical Channel Acknowledge

H.245 (TCP Dynamic Port)

RTP Stream RTP Stream RTCP Stream

Media (UDP)

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Basic H.323 Call

Gatekeeper A
LRQ LCF ACF ACF

Gatekeeper B

RRQ/RCF ARQ

IP Network
H.225 (Q.931) Setup H.225 (Q.931) Alert and Connect H.245

RRQ/RCF

ARQ

V
Gateway A Phone A

RTP

V
Gateway B Phone B

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Deploying H.323 Networks

Minimizes GK configuration Addition of new zones Addition of new NPAs Addition of new rate centers

DGK

GK

GK

GK

LA GW #1

West Zone

LA GW #2

Chicago GW Midwest Zone

NY GW East Zone

Rate Intra-LATA Rate Center #1 Toll Center #1

Local PSTN

Local PSTN

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MGCP/H.248/MegacoArchitectures

Call Agent SS7

Call Agent

P S T N

IMT

PSTN
PRI Access Gateway

P S T N

MGCP / H.248 / Megaco RTP

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Deploying MGCP/H.248/Megaco Networks

OSS CA SS7 Backhaul SS7 SLT MG
MG MG

MGCP and ISDN Backhaul

Billing and Measurement Server

STP PSTN
IMTs

TDM Voice
VoIP ISDN/PRI

Service Provider's TDM Network Service Provider's Packet Network

NAS/VoIP

Traditional TDM Traffic Modem Dial-up Traffic

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SIP Architecture
I N T E L L I G E N T S E R V I C E S 4426_02_2002_c1
Application Services

eMail

LDAP

Oracle

XML

CPL CPL

3pcc

SIP Proxy, Registrar & Redirect Servers SIP SIP SIP SIP User Agents (UA) PSTN

CAS or PRI

RTP (Media) Legacy PBX

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SIP Signaling

PSTN SIP VoIP Network

Calling Party INVITE 100 Trying INVITE 100 Trying 180 Ringing 200 OK ACK 180 Ringing 200 OK ACK

PSTN

Called Party

SIP Signaling and SDP Signaling (UDP or TCP)

Signaling

Media (UDP)

Bearer Or Media

RTCP Stream

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SIP Servers/Services
Location Database SIP Servers/ Services Where is this name/phone#? 3xx Redirection They moved, try this address

Registrar

Redirect

REGISTER Here I am

SIP Proxy Proxied INVITE Ill handle it for you

INVITE I want to talk to another UA

SIP User Agents

SIP User Agents

SIP-GW
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Deploying SIP Networks

PSTN 312 PSTN 212

Chicago POP Central Zone

NY POP East Zone

IP Network

West Zone SF POP

PSTN 415
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Topics to Discuss
History of VoIP VoIPEarly Adopters VoIPStandards and Standards Bodies VoIPMaking Sense of the Protocols The Great Voice Myth VoIPProtocol Challenges Summary

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Voice Myths
Myths
Networks can only be built one way Networks will only use one protocol All networks will converge

Facts
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

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Topics to Discuss
History of VoIP VoIPEarly Adopters VoIPStandards and Standards Bodies VoIPMaking Sense of the Protocols The Great Voice Myth VoIPProtocol Challenges Summary

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Interconnecting VoIP Networks

3 2 H.3

SIP

?
MGCP H.248 Megaco
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Connecting VoIP to SS7/C7 Networks

IAM

H.323

H.225 Setup (ANI,DN) Proceeding H.245

MGCP

CRCX ACK SDP

SIP

INVITE ACK SDP

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VoIP Interworking Issues

Service interworking
E.g.: H.450 <-> SIP <-> MGCP

Media interworking
End-to-end codec negotiation

Bearer interworking
End-to-end fax, modem, DTMF

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VoIP Interworking
Bearer level
Modem (relay/passthru) Fax (relay/passthru) T.38 T.37 DTMF (relay/passthru)

Service translation issues

Call deflection Park/hold

Signal issues
SDP H.245

Media level
Codec (negotiation, selection)

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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)

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Passthru Simplified
Voice Gateway PCM G.711 DSP G.711

G.729

IP Cloud

Voice Gateway G.711 DSP PCM G.711

G.729
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What Is Modem Passthru?

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

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Modem Passthru (Cont.)

Modem tone detection (<= V.90) Switchover signaling No VAD EC off RTP payload redundancy (10ms packetization) RFC2198 (optional)

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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

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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

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Fax RelayT.38
T.30 UDP T.30

PSTN

PSTN

IP
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
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DTMF

What is DTMF Why is it required? and where is it used? How do you transport it in IP? DTMF implementation

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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

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DTMF Schemes with VoIP Protocols

H.323

MGCP, H.248, Megaco In-Band

SIP

In-Band

In-Band
Cisco RTP, H.245 Alphanum, H.245 Signal, AVT Tones RFC2833

In-Band

Out-ofBand

Cisco RTP, NSE, NTE,RFC2833

RFC2833

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Topics to Discuss
History of VoIP VoIPEarly Adopters VoIPStandards and Standards Bodies VoIPMaking Sense of the Protocols The Great Voice Myth VoIPProtocol Challenges Summary

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Summary
Understand the possibilities and the issues Avoid protocol/product based bias Decide on application Consider market and business drivers Deploy whats possible today Choose signaling protocol depending on services intended to be offered Many possibilitiesstay tuned
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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

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Reference URLs

ITU: IETF: SIP:

www.itu.org www.ietf.org 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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