Presented by Jonathan Cumming jonathan.cumming@dataconnection.

com

Basics of Protocols SIP / H.323 / MGCP

Basics of Protocols - SIP/H.323/MGCP

VoIP Concepts Call Control Protocols
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SIP H.323

Media Gateway Control Protocols

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MGCP and MEGACO/H.248

Summary and Questions

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VoIP Concepts
Routing Softswitch / Application Server

Media Server

IP Network

Media Gateway Controller Endpoints Call Signaling and Media Call Signaling Media Media Gateway Control
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Media Gateway Gateway to other networks (PSTN / GSM / IP)

Call Control vs. Media Gateway Control

Call Control SIP and H.323
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Peer-to-Peer protocol between call agents Provides distributed call control

Media Gateway Control MGCP, MEGACO/H.248

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Master-Slave protocol, where the Media Gateway (MG) is controlled by the Media Gateway Controller (MGC) Allows decomposition into Call Agent and Media Gateway
Improves scalability Enables applications to share expensive media processing equipment
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Call Control Function

Endpoint Registration Routing Admission Control Call Establishment and Teardown Media Negotiation

Media Gateway Control Function

Discovery Control
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Media Resources User Interface Local Processing

Monitoring
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User Events Usage

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Call Control Protocols SIP and H.323 Call Control Protocols

Session Initiation Protocol

IETF-based Developed from work on multi-party conferences Releases
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RFC 2543 RFC 3261

1999 2002 - Better scalability and resilience

The protocol chosen for next generation mobile and fixed networks (3GPP and IMS) Huge amount of work extending the protocol
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SIP Standards
RFC 3261 defines the core SIP protocol Many extensions defined in additional RFCs
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RFC 3262 RFC 3263 RFC 3264

Reliable Provisional Responses Server Location Offer / Answer model

Higher-level features and conformance sets defined by organizations including ETSI and MSF
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SIP Terminology
Endpoints are SIP User Agents (UA)
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User Agent Clients (UAC) send requests User Agent Servers (UAS) process requests and send responses

Proxies route messages

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They cannot generate new requests B2BUA (Back-to-Back User Agent) can generate new requests

Registrars record endpoint location

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Normally co-located with a proxy or redirect server

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SIP Architecture
sip.dataconnection.com INVITE sip:jrc@dataconnection.com + SDP offer INVITE sip:jrc@128.96.41.1 + offer 200 OK + answer

200 OK + SDP answer

User Agent

RTP

User Agent

SDP offer/answer to negotiate the media RTP/RTCP carries the media directly between the endpoints
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Example SIP Request

Method Destination Route for the Response Request Line Protocol Version Original Destination Unique local identifier

INVITE sip:jrc@128.96.41.1 SIP/2.0 To: Jonathan Cumming <sip:jrc@dataconnection.com> From: Caller <sip:caller@there.com>;tag=12345767 Via: SIP/2.0/UDP 215.124.155.153 Current location of the sender Call-ID: 123456@here.com Contact: <sip:caller@215.124.155.153> Route: <sip:callerproxy.com> CSeq: 1 INVITE Route for the Request Content-Type: application/sdp v=0 c=IN IP4 215.124.155.153 m=audio 3456 RTP/AVP0
Description of message body Local Connection Descriptor (in SDP format)
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Headers

Body

Example SIP Response

Protocol Version Status Code Description Remote identifier Status Line

SIP/2.0 200 OK To: Jonathan <sip:jrc@dataconnection.com>;tag=12398 From: Caller <sip:caller@there.com>;tag=12345767 Via: SIP/2.0/UDP 215.124.155.153 Call-ID: 123456@here.com Contact: <sip:jrc@128.96.41.1> CSeq: 1 INVITE Current location of called party Content-Type: application/sdp v=0 c=IN IP4 128.96.41.1 m=audio 3456 RTP/AVP0
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Headers

Body

SIP Extensibility
Easy to prototype
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Anyone can define private Methods and Headers Text helps self-documentation

Built-in compatibility mechanisms

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Unrecognized methods and headers ignored Extensions mandated using Require: Capabilities advertised using Supported: and Allow:
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Presence and IM (SIMPLE)

Standard SIP addressing and routing mechanisms Presence
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SUBSCRIBE and NOTIFY methods (SIP Events) XML body describes state

Instant Messaging
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Text through signaling (MESSAGE message) Separate media session (MSRP)

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H.323
Published by ITU-T Developed from H.320 - Conferencing over ISDN
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Adapted for unreliable packet-based networks.

Widely used for conferencing and IP telephony Releases

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H.323 H.323 H.323 H.323

v1 v2 v3 v4

1996 1998 - Useable VoIP support 1999 - Improved scalability 2000 - Improved web support, inc URLs
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H.323 Standards
H.323 is a collection of standards
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H.225 RAS Communication with gatekeepers H.225 Call Signaling Call Establishment H.225 Annex G Introduces Border Elements H.245 Multimedia Control Protocol

Related standards
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H.235 Security within H.245-based systems H.450 Supplementary Services

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H.225 RAS Registration, Admission and Status

Used between endpoint and gatekeeper to allow
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the gatekeeper to manage the endpoint the endpoint to request admission for a call the endpoint to resolve addresses

RAS messages
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Gatekeeper Discovery (Gxx), and Registration (Rxx) Admission (Axx), Location (Lxx), and Bandwidth (Bxx) Disengage (Dxx), Information (Ixx), and various others
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H.225 Call Signaling

Used to establish calls between entities Derived from Q.931 (ISDN) Example messages
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Setup Call Proceeding Alerting

LRQ Gatekeeper LCF Gatekeeper

ARQ ARQ ACF Setup

ACF

Telephone

Connect

Telephone
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H.245
Provides
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Terminal capability exchange, e.g. Codecs supported Channel signaling to open the media sessions Conference control

Optimization options
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H.245 can be tunneled in H.225 Call Signaling channel

Normally transmitted in its own TCP connection

Fast connect does not establish an H.245 channel

Relevant H.245 fields passed in H.225 Setup message
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Registration Compared
SIP
Discovery
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H.323
Discovery
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DHCP Multicast REGISTER

Registration
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GRQ -> Gatekeeper GCF returned on success GRJ on error

REGISTER -> Registrar 200 OK on success >= 300 on error

Registration
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RRQ -> Gatekeeper Returns RCF or RQJ

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Call Setup Compared

SIP
Routing
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H.323
Routing
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INVITE routed by proxies

Lxx (GK GK)

Admission control
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Admission control
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Imposed by proxies on path

Axx (Endpoint GK)

Call Setup
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Call Setup
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INVITE -> <- 180 Ringing <- 200 OK ACK ->

Setup -> <- Proceeding <- Connect Ack ->

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Comparison Summary
Similarities
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H.323 Advantages
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Use RTP and RTCP for media transport Support call routing through proxies/gatekeepers using username, phone numbers or URLs Similar flows

More compact messages More mature (in some areas)

SIP Advantages
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Easier to prototype More flexible extensibility More scalable

loop detection

Differences
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Encoding (Text vs. ASN.1) Standardized Feature sets

Conference control Attended and blind transfer Caller Preferences

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Same messages used throughout network Cleaner separation of layers Use of SDP is compatible with MGCP and MEGACO
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Media Gateway Control Protocols Call Control Protocols MGCP and MEGACO/H.248

MGCP
Designed to control PSTN access by IP terminals Developed by Cisco, Telcordia and Level 3 Published by IETF as Informational RFC 2705, Oct 1999 Widely used in cable networks (PacketCable standards)

MEGACO/H.248
Jointly developed by IETF and ITU as a replacement to MGCP Cleaner and more powerful architecture => simpler flows Most MGCP messages have direct equivalents in Megaco Mandated for next generation networks (inc. IMS), but not yet as widely deployed
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MGCP/MEGACO Overview
Master/Slave control of media resources Text encoding
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Binary coding available for MEGACO, but hardly used

Use SDP to describe the media Packages define extensions

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Many defined, e.g. Basic Line, ATM, DTMF

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MGCP/MEGACO Terminology
MCGP
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MEGACO
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Endpoint A media source or sink. A media gateway is considered as a collection of endpoints, e.g. DS0, Analog line, etc. Connection A connection is an association between two endpoints, which may be on the same or different MGs.

Termination A media source or sink. This could be either a physical device, e.g. DS0, or an ephemeral termination such as an RTP stream. Context A connection is created by placing terminations into the same context
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MGCP/MEGACO examples
Media bridging
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MGCP connects endpoints to each other MEGACO places terminations in the same context

Connection

Conference Bridge Context

Conferencing
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Ephemeral Termination

MCGP connects endpoints to a conference bridge MEGACO places terminations in the same context
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Example MGCP message

Transaction ID Command Type Endpoint Indentifier Protocol Version

CRCX
Local Connection Options

1204

trunk-group-1/17@whatever.net
Call ID

MGCP 1.0

C:A3C47F21456789F0 L:p:10, a:G.711 M:recvonly

Compression Algorithm

Packetization Period

Connection Mode
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Summary
SIP and H.323 are equivalent MCGP and MEGACO are equivalent
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Call and Media Gateway control are complementary

SIP and MEGACO are the protocols of the future

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SIP is easier to extend and develop with than H.323 MEGACO has a more powerful architecture that MGCP

But H.323 and MGCP will still be here in 10 years

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Data Connection Overview

Background
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Founded in 1981 Headquarters in Enfield, UK 300 employees across 7 worldwide locations

Portable network protocols

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VoIP - SIP, MGCP, Megaco/H.248, Session Border Controller MPLS - RVSP-TE, LDP, CR-LDP, VPN IP Routing - Unicast and Multicast ATM
- UNI, PNNI, IPOA, ILMI

Stable and Independent

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Steady, profitable growth Privately held & self-funded by Employee Benefit Trust No plans for IPO or sale

Messaging/Directory/Confercing MetaSwitch
$M
55 50 45 40 35 30 25 20 15 10 5 0

REV ENUE EARNINGS

World Class Team

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Data Connection Group Results, 1982-2004

Recruit, invest, retain Retention: <2% turnover

8 2 84 86 88 90 9 2 94 96 98 00 02 04

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Questions

Thank you for listening You can also contact me at Booth 206

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Session Description Protocol (SDP)

Used to negotiate media channels and codecs Text-based protocol defined by the IETF Used by SIP, MGCP and MEGACO Offer/Answer handshake
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Offer contains list of supported streams and codecs Answer contains list of accepted streams and codecs

Supports wide range of media: RTP, ATM, video

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