Tema 4 - Protocolos VoIP_Overview.pdf

Voice over IP Protocols

An Overview

Version 2 - March 9, 2001 2

RTCPRTP

SIP, H.323 and MGCPSIP, H.323 and MGCP

IP

MGCP

Call Control and Signaling Signaling and Gateway Control

Media

H.225

Q.931

H.323

H.323 Version 1 and 2 supports H.245 over TCP, Q.93 1 over TCP and RAS over UDP.

H.323 Version 3 and 4 supports H.245 over UDP/TCP a nd Q.931 over UDP/TCP and RAS over UDP.

SIP supports TCP and UDP.

TCP

RAS

UDP

SIPH.245

Audio/Video

RTSP


What is in this ModuleWhat is in this Module

Objectives:This module provides an introductory overview of th e voice over IP protocols: SIP, H.323 and MGCP. At the end of this module, you will:• Understand the basics of SIP and its architecture.• Understand H.323 and how it compares to SIP.• Understand MGCP.

Target Audience:Marketing or business development professional who would like anintroductory yet technical overview of the voice ov er IP protocols.

Module Title:Voice over IP Protocol – An Overview

Session Initiation Protocol


What is SIP?What is SIP?

“

”

Session Initiation Protocol - An application layer signaling protocol that defines initiation, modification and termination of interactive, multimedia communication sessions between users.

IETF RFC 2543 Session Initiation Protocol


SIP FrameworkSIP Framework

• Session initiation.

• Multiple users.

• Interactive multimedia applications.


Redirect Server

SIP Distributed ArchitectureSIP Distributed Architecture

Location Server

Registrar Server

User Agent

Proxy Server

Gateway

PSTN

SIP Components

Proxy Server


User AgentsUser Agents

An application that initiates, receives and terminates calls. • User Agent Clients (UAC) – An entity that

initiates a call.

• User Agent Server (UAS) – An entity that receives a call.

�Both UAC and UAS can terminate a call.


Proxy ServerProxy Server

• An intermediary program that acts as both a server and a client to make requests on behalf of other clients.

• Requests are serviced internally or by passing them on, possibly after translation, to other servers.

• Interprets, rewrites or translates a request message before forwarding it.


Location ServerLocation Server

• A location server is used by a SIP redirect or proxy server to obtain information about a called party’s possible location(s).


Redirect ServerRedirect Server

• A server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client.

• Unlike a proxy server, the redirect server does not initiate its own SIP request.

• Unlike a user agent server, the redirect server does not accept or terminate calls.


Registrar ServerRegistrar Server

• A server that accepts REGISTER requests.

• The register server may support authentication.

• A registrar server is typically co -located with a proxy or redirect server and may offer location services.


SIP Messages – Methods and Responses

SIP Messages – Methods and Responses

SIP Methods:• INVITE – Initiates a call by

inviting user to participate in session.

• ACK - Confirms that the client has received a final response to an INVITE request.

• BYE - Indicates termination of the call.

• CANCEL - Cancels a pending request.

• REGISTER – Registers the user agent.

• OPTIONS – Used to query the capabilities of a server.

• INFO – Used to carry out-of-bound information, such as DTMF digits.

SIP Responses:• 1xx - Informational Messages.• 2xx - Successful Responses.• 3xx - Redirection Responses.• 4xx - Request Failure

Responses.• 5xx - Server Failure Responses.• 6xx - Global Failures

Responses.

SIP components communicate by exchanging SIP messag es:


SIP HeadersSIP Headers

• SIP borrows much of the syntax and semantics from H TTP.

• A SIP messages looks like an HTTP message – message formatting, header and MIME support.

• An example SIP header:-----------------------------------------------------------------

SIP Header

-----------------------------------------------------------------

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.6.21:5060

From: sip:[email protected]

To: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 100 INVITE

Expires: 180

User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled

Accept: application/sdp

Contact: sip:[email protected]:5060

Content-Type: application/sdp


SIP AddressingSIP Addressing

• The SIP address is identified by a SIP URL, in the format: user@host.

• Examples of SIP URLs:

–sip:[email protected]




Process for Establishing Communication


Establishing communication using SIP usually occurs in six steps:

1. Registering, initiating and locating the user.2. Determine the media to use – involves delivering a

description of the session that the user is invited to.3. Determine the willingness of the called party to

communicate – the called party must send a response message to indicate willingness to communicate – acc ept or reject.

4. Call setup.5. Call modification or handling – example, call transf er

(optional).6. Call termination.


RegistrationRegistration

• Each time a user turns on the SIP user client (SIP IP Phone, PC, or other SIP device), the client registers with the proxy/registration server.

• Registration can also occur when the SIP user client needs to inform the proxy/registration server of its location.

• The registration information is periodically refreshed and each user client must re-register with the proxy/registration server.

• Typically the proxy/registration server will forward this information to be saved in the location/redirect server.

SIP Messages:REGISTER – Registers the address listed in the To header field.200 – OK.

Proxy/ Registration Server

SIP PhoneUser

Location/Redirect Server

REGISTER REGISTER

200200


Simplified SIP Call Setup and Teardown (1)


100 Trying F3

180 (Ringing) F8

RTP MEDIA PATH

BYE F13

200 (OK) F14

Call Teardown

MediaPath

Call Setup

INVITE F1

atlanta.comProxy

biloxi.comProxy

Bob´s SIP PhoneAlice´s softphone

100 Trying F5

INVITE F2INVITE F4

180 (Ringing) F6

180 (Ringing) F7200 (OK) F9

200 (OK) F10200 (OK) F11

ACK F12




INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhdsMax-Forwards: 70To: Bob <sip:[email protected]> From: Alice <sip:[email protected]>;tag=1928301774Call-ID: [email protected]: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 142

SIP/2.0 200 OK Via: SIP/2.0/UDP server10.biloxi.com

;branch=z9hG4bKnashds8;received=192.0.2.3Via: SIP/2.0/UDP bigbox3.site3.atlanta.com

;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2Via: SIP/2.0/UDP pc33.atlanta.com

;branch=z9hG4bK776asdhds ;received=192.0.2.1To: Bob <sip:[email protected]>;tag=a6c85cfFrom: Alice <sip:[email protected]>;tag=1928301774Call-ID: [email protected]: 314159 INVITEContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 131




302 (Moved Temporarily)

INVITE

200 (OK)200 (OK)

ACK

INVITE302

(Moved Temporarily)ACK

INVITE

180 (Ringing)180 (Ringing)180 (Ringing)

200 (OK)ACKACK ACK

RTP MEDIA PATH

BYEBYE BYE

200 (OK)200 (OK) 200 (OK)Call Teardown

MediaPath

Call Setup

INVITELocation/Redirect ServerProxy Server Proxy Server User AgentUser Agent

INVITE


SIP – Design FrameworkSIP – Design Framework

SIP was designed for:• Integration with existing IETF protocols.

• Scalability and simplicity.

• Mobility.

• Easy feature and service creation.


Integration with IETF Protocols (1)


Other IETF protocol standards can be used to build a SIP based application. SIP can works with existing IETF protocols, for example:

• RSVP - to reserve network resources.

• RTP Real Time Protocol -to transport real time data and provide QOS feedback.

• RTSP Real Time Streaming Protocol - for controlling delivery of streaming media.

• SAP Session Advertisement Protocol - for advertising multimedia session via multicast.




• SDP Session Description Protocol – for describing multimedia sessions.

• MIME – Multipurpose Internet Mail Extension –defacto standard for describing content on the Internet.

• HTTP – Hypertext Transfer Protocol - HTTP is the standard protocol used for serving web pages over the Internet.

• COPS – Common Open Policy Service.

• OSP – Open Settlement Protocol.


Session Description Protocol (SDP)

Session Description Protocol (SDP)

Sesion descripción

v= (protocol version)o= (owner/creator and sessionidentifier). s= (session name)i=* (session information)u=* (URI of description)e=* (email address)p=* (phone number)c=* (connection information)b=* (bandwidth information)z=* (time zone adjustments)k=* (encryption key)t= (time the session is active)r=* (zero or more repeat times)m= (media name and transportaddress)a=* (zero or more mediaAttribute lines)

Ej. SDP Descriptionv=0o=mhandley 28908445262890842807 IN IP4 126.16.64.4s=SDP Seminari=A Seminar on the sessiondescription protocolu=http://www.cs.ucl.ac.uk/staff/M.Handley/[email protected] (Mark Handley)c=IN IP4 224.2.17.12/127t=2873397496 2873404696a=recvonlym=audio 49170 RTP/AVP 0m=video 51372 RTP/AVP 31m=application 32416 udp wba=orient:portrait


ScalabilityScalability

• The SIP architecture is scalable, flexible and distributed.

–Functionality such as proxying, redirection, location, or registration can reside in different physical servers.

–Distributed functionality allows new processes to be added without affecting other components.


SimplicitySimplicity

SIP is designed to be:

• “Fast and simple in the core.”

• “Smarter with less volume at the edge.”

• Text based for easy implementation and debugging.


MobilityMobility

• SIP supports user mobility by proxying and redirecting requests to a user’s current location.

• The user can be using a PC at work, PC at home, wireless phone, IP phone, or regular phone.

• The user must register their current location.

• The proxy server will forward calls to the user’s current location.

• Example mobility applications include presence and call forking.


Feature CreationFeature Creation

A SIP based system can support rapid feature and service creations.

For example, features and services can be created using:• Call Processing Language (CPL).

• Common Gateway Interface (CGI).


Feature Creation (2)Feature Creation (2)

SIP can support these features and applications:

• Basic call features (call waiting, call forwarding, call blocking etc.).

• Unified messaging.

• Call forking.

• Click to talk.

• Presence.

• Instant messaging.

• Find me / Follow me.


References References

For more information on SIP refer to:• http://www.ietf.org/html.charters/sip -

charter.html

• http://www.cs.columbia.edu/sip/

• http://www.sipcenter.com/

H.323


What is H.323?What is H.323?

“

”

Describes terminals and other entities that provide multimedia communications services over Packet Based Networks (PBN) which may not provide a guaranteed Quality of Service. H.323 entities may provide real-time audio, video and/or data communications.

ITU-T Recommendation H.323 Version 4


Call establishment and teardown.Call establishment and teardown.

H.323 defines:• Call establishment and teardown.

• Audio visual or multimedia conferencing.


H.323 Components H.323 Components

Terminal Gateway

Packet Based

Networks

Multipoint Control Unit

Gatekeeper

Circuit Switched

Networks


H.323 TerminalsH.323 Terminals

H.323 terminals are client endpoints that must support:• H.225 call control signaling.

• H.245 control channel signaling.

• RTP/RTCP protocols for media packets.

• Audio codecs.

�Video codecs support is optional.


H.323 GatewayH.323 Gateway

A gateway provides translation:• For example, a gateway can provide

translation between entities in a packet switched network (example, IP network) and circuit switched network (example, PSTN network).

• Gateways can also provide transmission formats translation, communication procedures translation, H.323 and non -H.323 endpoints translations or codec translation.


H.323 GatekeepersH.323 Gatekeepers

Gatekeepers provide these functions:

• Address translation.

• Admission control.

• Bandwidth control.

• Zone management.

• Call control signaling (optional).

• Call authorization (optional).

• Bandwidth management (optional).

• Call management (optional).

Gatekeepers are optional but if present in a H.323 sys tem, all H.323 endpoints must register with the gatekeeper and receive permission before making a call.


H.323 Multipoint Control UnitH.323 Multipoint Control Unit

MCU provide support for conferences of three or more endpoints.

An MCU consist of:• Multipoint Controller (MC) – provides control

functions.

• Multipoint Processor (MP) – receives and processes audio, video and/or data streams.


H.323 is an “Umbrella” Specification

H.323 is an “Umbrella” Specification

Call Control and SignalingH.245 - Capabilities advertisement, media channel establishment, and conference control.

H.225

Q.931 - call signaling and call setup.

RAS - registration and other admission control with a gatekeeper.

Call Control and

Signaling

Data/FaxMedia

IP

UDP

RTP

Audio

Codec

G.711

G.723

G.729

Video

Codec

H.261

H.263RTCP

H.225

Q.931

H.225

RASH.245T.120 T.38

TCP TCPUDPTCP

Data/FaxT.120 – Data conferencing.T.38 – Fax.

Media H.261 and H.263 – Video codecs.G.711, G.723, G.729 – Audio codecs.RTP/RTCP – Media.

H.323


Other ITU H. Recommendation that work with H.323

Other ITU H. Recommendation that work with H.323

Protocol Description

H.235 Specifies security and encryption for H.323 and H.245 based terminals.

H.450.N H.450.1 specifies framework for supplementary services. H.450.Nrecommendation specifies supplementary services such as call transfer, call diversion, call hold, call park, call waiting, message waiting indication, name identification, call completion, call offer, and call intrusion.

H.246 Specifies internetworking of H Series terminals with circuit switched terminals.


H.323 Components and Signaling

H.323 Components and Signaling

H.245 – A protocol for capabilities advertisement, m edia channel establishment and conference control.

H.225 - Call Control.

- Q.931 – A protocol for call control and call setup.

- RAS – Registration, admission and status protocol u sed for communicating between an H.323 endpoint and a gatekeeper.

PSTN

Gatekeeper

Terminal

H.225/RAS messages over RAS channel

GatewayH.245 messages over call control channel

H.225/Q.931 messages over call signaling channel

H.225/RAS messages over RAS channel

H.225/Q.931 (optional) H.225/Q.931 (optional)

H.245 messages (optional) H.245 messages (optional)




Establishing communication using H.323 may occurs in five steps:1. Call setup.

2. Initial communication and capabilities exchange.

3. Audio/video communication establishment.

4. Call services.

5. Call termination.


Simplified H.323 Call SetupSimplified H.323 Call Setup

• Both endpoints have previously registered with the gatekeeper.

• Terminal A initiate the call to the gatekeeper. (RAS messages are exchanged).

• The gatekeeper provides information for Terminal A to contact Terminal B.

• Terminal A sends a SETUP message to Terminal B.

• Terminal B responds with a Call Proceeding message and also contacts the gatekeeper for permission.

• Terminal B sends a Alerting and Connect message.

• Terminal B and A exchange H.245 messages to determine master slave, terminal capabilities, and open logical channels.

• The two terminals establish RTP media paths.

Terminal A Gatekeeper Terminal B

RAS messagesCall Signaling Messages

1. ARQ2. ACF

5. ARQ6. ACF

3. SETUP4. Call Proceeding

7.Alerting8.Connect

H.245 MessagesRTP Media Path

Note: This diagram only illustrates a simple point-to-point call setup where call signaling is not routed to the gatekeeper. Refer to the H.323 recommendation for more call setup scenarios.


Registration, Admission, and Status – RAS

Registration, Admission, and Status – RAS

T GKRegistrationRequest (RRQ)

RegistrationConfirm (RCF)

Admission Request (ARQ)(endpoint is registered)

Admission Confirmation (ACF)(endpoint may place call)

DisengageRequest (DRQ)

Disengage Confirm (DCF)(call has terminated)

T Terminal

GK Gatekeeper

GW Gateway

Symbol Key:


H.225.0 Call SignalingH.225.0 Call Signaling

Allows an endpoint to initiate and terminate a call with another endpoint

GWSetup

Alerting

Connect

(call is established)

Release Complete(call is terminated)

H.245

Signaling

may take

place at

any point

T


H.225.0 Call Signaling Message

H.225.0 Call Signaling Message

TPKT Q.931 Header IE IE IE IE IE IE IE IE IE IE IE IE IE IE IE IE IE IE IE IE IE UUIE. . .

Four octets that separate messages on the wire (necessary for TCP). They are defined in section 6 of RFC

1006. There are 0x03, 0x00, HH, LL. HH and LL represent the entire message length, including the

TPKT header, in network byte order.

All messages have a Q.931 header that includes a single octet called the “protocol discriminator”

(0x08), three octets for the CRV (0x02, HH, LL, where 0x02 is the length of the CRV and HH

and LL are the two octets of the CRV in network byte order), and single octet for the message

type (specified in respective sections in Q.931).

Various Information Elements (IEs) that are appropriate for the message type. These

are listed in H.225.0, but note that any valid Q.931 IE may be transmitted and should

not result in a protocol failure by the endpoint.

The UUIE refer to the “User-User Information Element”. It should be

the last octet in the chain, but some implementations do not properly

order IEs. It is comprised of 0x7E, HH, LL, PD, and DATA. 0x7E is

the identifier for the User-User IE, HH and LL are the length of DATA

in network byte order, PD is a protocol discriminator for ASN.1 (0x05)

and DATA is the ASN.1 PER encoded “H323-UserInformation”.


H.225.0 Information ElementsH.225.0 Information Elements

• Information elements carry additional information related to the specific message

• For example, SETUP has, among other things, a “Calling Party Number” IE, “Called Party Number” IE, “Display” IE, etc.

• Every H.225.0 message has a UUIE, though this is not true of Q.931

• H.225.0 made a number of changes to Q.931 and should be the guiding document


Open a

channel in

each

direction

H.245 SignalingH.245 Signaling

H.245 is used to negotiate capabilities and to control aspects of the conference between two or more endpoints

GWTCS + MSD

TCS + TCS Ack + MSD Ack

TCS Ack + MSD Ack + OLC

OLC Ack + OLC

OLC Ack

T


H.245 MessageH.245 Message

TPKT H.245 PDU

Four octets that separate messages on the wire (necessary for TCP). They are defined in section 6 of RFC

1006. There are 0x03, 0x00, HH, LL. HH and LL represent the entire message length, including the

TPKT header, in network byte order.

H.245 messages are encoded in ASN.1 PER and follow the TPKT header in the H.245 Control

Channel.

Additional H.245 PDUs may be encoded following the first one. However, many

implementations cannot handle this and, as such, it is ill-advised to place them end-to-

end like this. It is strongly recommended to place only one between each TPKT

header, but do be prepared for the case that more than one PDU does exist following

TPKT

H.245 PDU H.245 PDU H.245 PDU. . .


H.245 Signaling. Capabilities Exchange

H.245 Signaling. Capabilities Exchange

• The capability exchange (or “caps exchange”) allows two endpoints to exchange information about what media capabilities they possess, such as G.711, G.723, H.26 1, and H.263

• Along with the type of media, specific details abou t the maximum number of audio frames or samples per packet i s exchanged, information about support for silence supp ression (VAD), etc. are exchanged

• Using this capability information, endpoints can sel ect preferred codes that are suitable to both parties

• The terminalCapabilitySet (TCS) must be the first mess age transmitted on the H.245 Control Channel


H.245 Signaling. Master Slave Determination

H.245 Signaling. Master Slave Determination

• Once capabilities are exchanged, the endpoints negotiate master and slave roles (MSD)

Actually the master/slave messages can be sent alon g with the TCS message

• The master in a point to point conference really on ly has the power to indicate when channels are in conflict (e.g., when one the other terminal tries t o open a channel that is not compatible)

• The slave device must yield to the requests of the master device and reconfigure channels appropriately


H.245 Signaling. Logical Channel Signaling (1)


• Channels are opened by exchanging “openLocalChannel” (OLC) messages

• The OLC will contain one of the capabilities that was previously advertised by the other endpoint

• Voice and video channels are “unidirectional”, so each end must transmit an OLC to open a logical channel




• Within the OLC, a “session ID” is assigned

• Session 1 is the default audio session, 2 is the default video session, and 3 is the default data session

• Additional session IDs may be used, but are assigned by the master in the call

• There is a relationship between H.245 sessions IDs and RTP: OLCs with the same session ID are considered to be part of the same RTP/RTCP session


Versions of H.323Versions of H.323

Version Reference for key feature summary

H.323 Version 3 http://www.packetizer.com/ipmc/h323/whatsnew_v3.html

Date

H.323 Version 1 New release. Refer to the specification.

http://www.packetizer.com/ipmc/h323/

May 1996

H.323 Version 2 http://www.packetizer.com/ipmc/h323/whatsnew_v2.html

January 1998

September 1999

H.323 Version 4 November 2000 http://www.packetizer.com/ipmc/h323/whatsnew_v4.html

H.323 Version 5 July 2003 http://www.packetizer.com/ipmc/h323/whatsnew_v5.html

H.323 Version 6 June 2006 http://www.packetizer.com/ipmc/h323/whatsnew_v6.html



For more information on H.323 refer to:ITU-T• http://www.itu.int/itudoc/itu-t/rec/index.html

Packetizer• http://www.packetizer.com/ipmc/h323/

Open H.323• http://www.openH323.org

SIP and H.323

Comparing


Comparing SIP and H.323 -Similarities

Comparing SIP and H.323 -Similarities

Functionally, SIP and H.323 are similar. Both SIP and H.323 provide:

• Call control, call setup and teardown.

• Basic call features such as call waiting, call hold, call transfer, call forwarding, call return, call identification, or call park.

• Capabilities exchange.


Comparing SIP and H.323 -Strengths

Comparing SIP and H.323 -Strengths

• H.323 – Defines sophisticated multimedia conferencing. H.323 multimedia conferencing can support applications such as whiteboarding, data collaboration, or video conferencing.

• SIP – Supports flexible and intuitive feature creati on with SIP using SIP-CGI (SIP-Common Gateway Interface) and CPL (Call Processing Language).

• SIP – Third party call control is currently only available in SIP. Work is in progress to add this functionality to H.323.


Table 1 - SIP and H.323

H.323SIP

ITU.IETF.

Peer-to-Peer. Peer-to-Peer.

Telephony based. Borrows call signaling protocol from ISDN Q.SIG.

Internet based and web centric. Borrows syntax and messages from HTTP.

Intelligent H.323 terminals.Intelligent user agents.

H.323 Gatekeeper.SIP proxy, redirect, location, and registration servers.

Widespread.Interoperability testing between various vendor’s products is ongoing at SIP bakeoffs.

SIP is gaining interest.

Information

Standards Body

Relationship

Origins

Client

Core servers

Current Deployment

Interoperability IMTC sponsors interoperability events among SIP, H.323, and MGCP. For more information, visit: http://www.imtc.org/


Table 2 - SIP and H.323Table 2 - SIP and H.323

Information H.323SIP

Capabilities Exchange

Supported by H.245 protocol. H.245 provides structure for detailed and precise information on terminal capabilities.

SIP uses SDP protocol for capabilities exchange. SIP does not provide as extensive capabilities exchange as H.323.

Control Channel Encoding Type

Binary ASN.1 PER encoding.Text based UTF-8 encoding.

Server Processing

Version 1 or 2 – Stateful.

Version 3 or 4 – Stateless or stateful.

Stateless or stateful.

Quality of Service

Bandwidth management/control and admission control is managed by the H.323 gatekeeper.

The H323 specification recommends using RSVP for resource reservation.

SIP relies on other protocols such as RSVP, COPS, OSP to implement or enforce quality of service.


Table 3 - SIP and H.323Table 3 - SIP and H.323


Security Registration - If a gatekeeper is present, endpoints register and request admission with the gatekeeper.

Authentication and Encryption -H.235 provides recommendations for authentication and encryption in H.323 systems.

Registration - User agent registers with a proxy server.

Authentication - User agent authentication uses HTTP digest or basic authentication.

Encryption - The SIP RFC defines three methods of encryption for data privacy.

Endpoint Location and Call Routing

Uses E.164 or H323ID alias and a address mapping mechanism if gatekeepers are present in the H.323 system.

Gatekeeper provides routing information.

Uses SIP URL for addressing.

Redirect or location servers provide routing information.


Table 4 – SIP and H.323Table 4 – SIP and H.323


Features Basic call features.Basic call features.

Conferencing Basic conferencing without conference or floor control.

Comprehensive audiovisual conferencing support.

Data conferencing or collaboration defined by T.120 specification.

Service or Feature Creation

Supports flexible and intuitive feature creation with SIP using SIP-CGI and CPL.

Some example features include presence, unified messaging, or find me/follow me.

H.450.1 defines a framework for supplementary service creation.

Note: Basic call features include: call hold, call waiting, call transfer, call forwarding, caller identification, and call park.


ReferenceReference

This section cites a document that provides a comprehensive comparison on H.323 and SIP:

Nortel Networks3GPP S2, Tokyo Japan, S2-000505

http://www.cs.columbia.edu/sip/papers.html

MGCP

Media Gateway Control Protocol


What is MGCP?What is MGCP?

“

”

Media Gateway Control Protocol - A protocol for controlling telephony gateways from external call control elements called media gateway controllers or call agents.

IETF RFC 2705 Media Gateway Control Protocol


ComponentsComponents

Call agent or media gateway controller• Provides call signaling, control

and processing intelligence to the gateway.

• Sends and receives commands to/from the gateway.

Gateway• Provides translations between

circuit switched networks and packet switched networks.

• Sends notification to the call agent about endpoint events.

• Execute commands from the call agents.

Call Agent or Media Gateway

Controller(MGC)

Call Agent or Media Gateway

Controller(MGC)

SIPH.323

MGCP MGCP

Media Gateway(MG)

Media Gateway(MG)


Simplified Call FlowSimplified Call Flow

• When Phone A goes offhook Gateway A sends a signal to the call agent.

• Gateway A generates dial tone and collects the dialed digits.

• The digits are forwarded to the call agent.

• The call agent determines how to route the call.

• The call agent sends commands to Gateway B.

• Gateway B rings phone B.• The call agent sends

commands to both gateways to establish RTP/RTCP sessions.

Gateway A Gateway B

AnalogPhone A

AnalogPhone B

Call AgentMedia Gateway Controller

MGCP MGCP

RTP/RTCP


MGCP CommandsMGCP Commands

Call Agent Commands:• EndpointConfiguration

• NotificationRequest

• CreateConnection

• ModifyConnection

• DeleteConnection

• AuditEndpoint

• AuditConnection

Gateway Commands:• Notify

• DeleteConnection

• RestartInProgress


Characteristics of MGCPCharacteristics of MGCP

MGCP:

• A master/slave protocol.

– Assumes limited intelligence at the edge (endpoints) and intelligence at the core (call agent).

– Used between call agents and media gateways.

– Differs from SIP and H.323 which are peer-to-peer protocols.

• Interoperates with SIP and H.323.


MGCP, SIP and H.323MGCP, SIP and H.323

• MGCP divides call setup/control and media establishment functions.

• MGCP does not replace SIP or H.323. SIP and H.323 provide symmetrical or peer-to-peer call setup/control.

• MGCP interoperates with H.323 and SIP. For example,

– A call agent accepts SIP or H.323 call setup requests.

– The call agent uses MGCP to control the media gateway.

– The media gateway establishes media sessions with other H.323 or SIP endpoints.

Call Agent/Media

Gateway Controller

Media Gateway

MGCP

H.323 Gateway

H.323Gateway

H.323

Media RTP/RTCP

In this example, an H.323 gateway is “decomposed” into:

–A call agent that provides signaling.

–A gateway that handles media.

–MGCP protocol is used to control the gateway.


Example ComparisonExample Comparison

MGCP1. A user picks up analog phone and dials a

number.

2. The gateway notifies call agent of the phone (endpoint) event.

3. The Call agent determines capabilities, routing information, and issues a command to the gateways to establish RTP/RTCP session with other end.

H.323Gateway

H.323Gateway

AnalogPhone

AnalogPhone

Gateway A Gateway B

AnalogPhone

Call Agent/Media

Gateway Controller

RTP/RTCP

AnalogPhone

H.323 1. A user picks up analog phone and

dials a number.

2. The gateway determines how to route the call.

3. The two gateways exchange capabilities information.

4. The terminating gateway rings the phone.

5. The two gateways establish RTP/RTCP session with each other.

5.RTP/RTCP1

3

4

1

2


What is Megaco?What is Megaco?

A protocol that is evolving from MGCP and developed jointly by ITU and IETF:

• Megaco - IETF.

• H.248 or H.GCP - ITU.

For more information refer to:

• IETF - http://www.ietf.org/html.charters/megaco-charter.html

• Packetizer - http://www.packetizer.com/ipmc/h248/



For more information on MGCP refer to:IETF• http://www.ietf.org/rfc/rfc2705.txt?number=2705

Summary


SummarySummary

• SIP and H.323 are comparable protocols that provide call setup, call teardown, call control, capabilities exchange, and supplementary features.

• MGCP is a protocol for controlling media gateways from call agents. In a VoIP system, MGCP can be used with SIP or H.323. SIP or H.323 will provide the call control functionality and MGCP can be used to manage media establishment in media gateways.

Additional References


General VoIP ReferenceGeneral VoIP Reference

Pulver – IP Telephony News• http://www.pulver.com

Internet Telephony• http://www.internettelephony.com

An overview poster of the SIP, MGCP, and H323 protocols.• http://www.protocols.com/

Date post:	03-Dec-2015
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