Tema 4 - Protocolos VoIP - Overview PDF

Voice over IP Protocols
An Overview
SIP, H.323 and MGCP

Call Control and Signaling
Signaling and
Gateway Control
Media
Audio/
Video
H.323
H.225
H.245
Q.931
RAS
SIP
MGCP
TCP
RTP
RTCP
RTSP
UDP
IP
H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP.
H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP.
SIP supports TCP and UDP.
Version 2 - March 9, 2001
What is in this Module

Module Title:
Voice over IP Protocol An Overview
Objectives:
This module provides an introductory overview of the 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 an
introductory yet technical overview of the voice over IP protocols.
Session Initiation Protocol
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 Framework
Session initiation.
Multiple users.
Interactive
multimedia
applications.
SIP Distributed Architecture

SIP Components
Location
Server
Redirect
Server
Registrar
Server
PSTN
User Agent
Gateway
Proxy
Server
Proxy
Server
User 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 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 Server
A location server is used by a SIP redirect or
proxy server to obtain information about a
called partys possible location(s).
10
Redirect 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.
11
Registrar 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.
12
SIP Messages Methods and

Responses
SIP components communicate by exchanging SIP messages:
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-ofbound 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.
13
SIP Headers
SIP borrows much of the syntax and semantics from HTTP.
A SIP messages looks like an HTTP message message formatting,
header and MIME support.
An example SIP header:
----------------------------------------------------------------SIP Header
----------------------------------------------------------------INVITE sip:5120@192.168.36.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.21:5060
From: sip:5121@192.168.6.21
To: <sip:5120@192.168.36.180>
Call-ID: c2943000-e0563-2a1ce-2e323931@192.168.6.21
CSeq: 100 INVITE
Expires: 180
User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled
Accept: application/sdp
Contact: sip:5121@192.168.6.21:5060
Content-Type: application/sdp
14
SIP Addressing
The SIP address is identified by a SIP URL, in
the format: user@host.
Examples of SIP URLs:
sip:hostname@vovida.org
sip:hostname@192.168.10.1
sip:14083831088@vovida.org
15
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 accept
or reject.
4. Call setup.
5. Call modification or handling example, call transfer
(optional).
6. Call termination.
16
Registration
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.
Proxy/
Registration
Server
SIP Phone
User
Location/
Redirect
Server
REGISTER
REGISTER
200
200
SIP Messages:
REGISTER Registers the address listed in the To
header field.
200 OK.
17
Simplified SIP Call Setup and

Teardown (1)
Alices softphone
Bobs SIP Phone
biloxi.com
atlanta.com
Proxy
Proxy
INVITE F1
INVITE F2
INVITE F4
100 Trying F3
100 Trying F5
Call
Setup
180 (Ringing) F6
180 (Ringing) F7
180 (Ringing) F8
200 (OK) F9
200 (OK) F10
200 (OK) F11

ACK F12
Media
Path
Call
Teardown
RTP MEDIA PATH

BYE F13
200 (OK) F14
18

Teardown (2)
INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds
Max-Forwards: 70
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@pc33.atlanta.com
CSeq: 314159 INVITE
Contact: <sip:alice@pc33.atlanta.com>
Content-Length: 142
SIP/2.0 200 OK
Via: SIP/2.0/UDP server10.biloxi.com
;branch=z9hG4bKnashds8;received=192.0.2.3
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com
;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2
Via: SIP/2.0/UDP pc33.atlanta.com
;branch=z9hG4bK776asdhds ;received=192.0.2.1
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@pc33.atlanta.com
CSeq: 314159 INVITE
Contact: <sip:bob@192.0.2.4>
Content-Length: 131
19

Teardown (3)
Proxy Server
User Agent
INVITE
Location/Redirect Server
INVITE
302
(Moved Temporarily)
User Agent
Proxy Server
ACK
INVITE
INVITE
302
(Moved Temporarily)
ACK
Call
Setup
180 (Ringing)
200 (OK)
ACK
Media
Path
Call
Teardown
180 (Ringing)
200 (OK)
ACK
INVITE
180 (Ringing)
200 (OK)
ACK
RTP MEDIA PATH

BYE
BYE
BYE
200 (OK)
200 (OK)
200 (OK)
20
SIP Design Framework

SIP was designed for:
Integration with existing IETF protocols.
Scalability and simplicity.
Mobility.
Easy feature and service creation.
21
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.
22
Integration with IETF

Protocols (2)
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.
23
Session Description Protocol

(SDP)
Sesion descripcin
Ej. SDP Description
v= (protocol version)
o= (owner/creator and session
identifier).
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 transport
address)
a=* (zero or more media
Attribute lines)
v=0
o=mhandley 2890844526
2890842807 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session
description protocol
u=http://www.cs.ucl.ac.uk/st
aff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 49170 RTP/AVP 0
m=video 51372 RTP/AVP 31
m=application 32416 udp wb
a=orient:portrait
24
Scalability
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.
25
Simplicity
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.
26
Mobility
SIP supports user mobility by proxying and
redirecting requests to a users 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
users current location.
Example mobility applications include
presence and call forking.
27
Feature 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).
28
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.
29
References
For more information on SIP refer to:
http://www.ietf.org/html.charters/sipcharter.html
http://www.cs.columbia.edu/sip/
http://www.sipcenter.com/
30
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
32
Call establishment and teardown.

H.323 defines:
Call establishment and teardown.
Audio visual or multimedia conferencing.
33
H.323 Components
Gatekeeper
Terminal
Multipoint
Control Unit
Packet Based
Circuit Switched
Networks
Networks
Gateway
34
H.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.
35
H.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.
36
H.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 system, all
H.323 endpoints must register with the gatekeeper and receive
permission before making a call.
37
H.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.
38
H.323 is an Umbrella
Specification
Media
H.323
H.261 and H.263 Video codecs.

G.711, G.723, G.729 Audio codecs.
RTP/RTCP Media.
Media
Data/Fax
Call Control and

Signaling
Data/Fax
T.120 Data conferencing.
T.38 Fax.
Audio
Codec
G.711
G.723
Call Control and Signaling

H.245 - Capabilities advertisement, media
channel establishment, and conference
control.
H.225
Q.931 - call signaling and call setup.
G.729
Video
Codec
H.261
RTCP
T.120 T.38
H.263
H.225 H.225
Q.931
RAS
TCP
UDP
H.245
RTP
UDP
TCP
TCP
IP
RAS - registration and other admission

control with a gatekeeper.
39
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.N

recommendation 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.
40
H.323 Components and

Signaling
H.225/RAS messages
over RAS channel
H.225/Q.931 (optional)
H.225/RAS messages
over RAS channel
Gatekeeper
H.225/Q.931 (optional)
H.245 messages (optional)
H.245 messages (optional)

H.225/Q.931 messages over
call signaling channel
Terminal
H.245 messages over

call control channel
PSTN
Gateway
H.245 A protocol for capabilities advertisement, media 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 used for communicating between an
H.323 endpoint and a gatekeeper.
41
Process for Establishing

Communication
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.
42
Simplified 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
1. ARQ
2. ACF
3. SETUP
4. Call Proceeding
5. ARQ
6. ACF
7.Alerting
8.Connect
H.245 Messages
RTP Media Path
RAS messages
Call Signaling Messages
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.
43
Registration, Admission, and

Status RAS
RegistrationRequest (RRQ)
T
RegistrationConfirm (RCF)
GK
(endpoint is registered)
Admission Request (ARQ)

Admission Confirmation (ACF)
(endpoint may place call)
DisengageRequest ( DRQ)
Symbol Key:
Disengage Confirm (DCF)
(call has terminated)
GK
Gatekeeper
GW
Gateway
Terminal
44
H.225.0 Call Signaling

Allows an endpoint to initiate and
terminate a call with another endpoint
Setup
Alerting
Connect
(call is established)
GW
H.245
Signaling
may take
place at
any point
Release Complete
(call is terminated)
45
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
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.
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.
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).
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.
46
H.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
47
H.245 Signaling
H.245 is used to negotiate capabilities
and to control aspects of the
conference between two or more
endpoints
T
TCS + MSD
TCS + TCS Ack + MSD Ack
TCS Ack + MSD Ack + OLC
OLC Ack + OLC
OLC Ack
GW
Open a
channel in
each
direction
48
H.245 Message
TPKT H.245 PDU H.245 PDU
...
H.245 PDU H.245 PDU
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-toend 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 messages are encoded in ASN.1 PER and follow the TPKT header in the H.245 Control
Channel.
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.
49
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.261, and
H.263
Along with the type of media, specific details about the
maximum number of audio frames or samples per packet is
exchanged, information about support for silence suppression
(VAD), etc. are exchanged
Using this capability information, endpoints can select
preferred codes that are suitable to both parties
The terminalCapabilitySet (TCS) must be the first message
transmitted on the H.245 Control Channel
50
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 along
with the TCS message
The master in a point to point conference really only

has the power to indicate when channels are in
conflict (e.g., when one the other terminal tries to
open a channel that is not compatible)
The slave device must yield to the requests of the
master device and reconfigure channels
appropriately
51
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
52
H.245 Signaling. Logical

Channel Signaling (2)
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
53
Versions of H.323
Version
H.323 Version 1
Date
May 1996
Reference for key feature summary

New release. Refer to the specification.
http://www.packetizer.com/ipmc/h323/
H.323 Version 2
January 1998
http://www.packetizer.com/ipmc/h323/whatsnew
_v2.html
H.323 Version 3
September 1999
_v3.html
H.323 Version 4
November 2000
_v4.html
H.323 Version 5
July 2003
_v5.html
H.323 Version 6
June 2006
_v6.html
54
References
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
55
Comparing
SIP and H.323
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.
57
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 creation
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.
58
Table 1 - SIP and H.323

Information
SIP
H.323
Standards Body
IETF.
ITU.
Relationship
Peer-to-Peer.
Peer-to-Peer.
Origins
Internet based and web centric.

Borrows syntax and messages
from HTTP.
Telephony based. Borrows call

signaling protocol from ISDN
Q.SIG.
Client
Intelligent user agents.
Intelligent H.323 terminals.
Core servers
SIP proxy, redirect, location, and

registration servers.
H.323 Gatekeeper.
Current
Deployment
Interoperability testing between

various vendors products is
ongoing at SIP bakeoffs.
Widespread.
SIP is gaining interest.

Interoperability
IMTC sponsors interoperability events among SIP, H.323, and MGCP.

For more information, visit: http://www.imtc.org/
59

Information
SIP
H.323
Capabilities
Exchange
SIP uses SDP protocol for

capabilities exchange. SIP does
not provide as extensive
capabilities exchange as H.323.
Supported by H.245 protocol.

H.245 provides structure for
detailed and precise information
on terminal capabilities.
Control
Channel
Encoding Type
Text based UTF-8 encoding.
Binary ASN.1 PER encoding.
Server
Processing
Stateless or stateful.
Version 1 or 2 Stateful.
Quality of
Service
SIP relies on other protocols such

as RSVP, COPS, OSP to
implement or enforce quality of
service.
Version 3 or 4 Stateless or
stateful.
Bandwidth management/control
and admission control is managed
by the H.323 gatekeeper.
The H323 specification
recommends using RSVP for
resource reservation.
60

Information
Security
SIP
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 SIP URL for addressing.

Redirect or location servers
provide routing information.
H.323
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.
Uses E.164 or H323ID alias and a
address mapping mechanism if
gatekeepers are present in the
H.323 system.
Gatekeeper provides routing
information.
61
Table 4 SIP and H.323

Information
SIP
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.
H.450.1 defines a framework for

supplementary service creation.
Some example features include

presence, unified messaging, or
find me/follow me.
Note: Basic call features include: call hold, call waiting, call transfer, call
forwarding, caller identification, and call park.
62
Reference
This section cites a document that provides a
comprehensive comparison on H.323 and SIP:
Nortel Networks
3GPP S2, Tokyo Japan, S2-000505
http://www.cs.columbia.edu/sip/papers.html
63
MGCP
Media Gateway Control
Protocol
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
65
Components
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)
SIP
H.323
Call Agent or
Media Gateway
Controller
(MGC)
MGCP
MGCP
Media Gateway
(MG)
Media Gateway
(MG)
66
Simplified 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.
Call Agent
Media Gateway Controller
MGCP
MGCP
RTP/RTCP
Gateway A
Analog
Phone A
Gateway B
Analog
Phone B
67
MGCP Commands
Call Agent
Commands:
Gateway Commands:
Notify
EndpointConfiguration
DeleteConnection
NotificationRequest
RestartInProgress
CreateConnection
ModifyConnection
DeleteConnection
AuditEndpoint
AuditConnection
68
Characteristics 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.
69
MGCP, SIP and H.323

MGCP divides call setup/control
and media establishment
functions.
In this example, an H.323 gateway is

decomposed into:
MGCP does not replace SIP or

H.323. SIP and H.323 provide
symmetrical or peer-to-peer call
setup/control.
A gateway that handles media.
A call agent that provides signaling.
MGCP protocol is used to control the

gateway.
H.323 Gateway
H.323
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
H.323
Gateway
MGCP
Media RTP/RTCP
Media
Gateway
70
Example Comparison
H.323
MGCP
1.
A user picks up analog phone and

dials a number.
1.
A user picks up analog phone and dials a

number.
2.
The gateway determines how to

route the call.
2.
The gateway notifies call agent of the phone

(endpoint) event.
3.
The two gateways exchange

capabilities information.
3.
4.
The terminating gateway rings the

phone.
The Call agent determines capabilities,

routing information, and issues a command
to the gateways to establish RTP/RTCP
session with other end.
5.
The two gateways establish

RTP/RTCP session with each other.
3
5.RTP/
RTCP
H.323
Gateway
Analog
Phone
4
H.323
Gateway
Call Agent/
Media
Gateway
Controller
1
Analog
Phone
Gateway A
Analog
Phone
RTP/
RTCP Gateway B
Analog
Phone
71
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/megacocharter.html
Packetizer - http://www.packetizer.com/ipmc/h248/
72
References
For more information on MGCP refer to:
IETF
http://www.ietf.org/rfc/rfc2705.txt?number=2705
73
Summary
Summary
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.
75
Additional References
General 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/
77

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Voice over IP Protocols

SIP, H.323 and MGCP

What is in this Module

Version 2 - March 9, 2001

Session Initiation Protocol

Session Initiation Protocol - An

Version 2 - March 9, 2001

Version 2 - March 9, 2001

SIP Distributed Architecture

Version 2 - March 9, 2001

Version 2 - March 9, 2001

Version 2 - March 9, 2001

Version 2 - March 9, 2001

Version 2 - March 9, 2001

Version 2 - March 9, 2001

SIP Messages Methods and

1xx - Informational Messages.

Version 2 - March 9, 2001

Version 2 - March 9, 2001

Process for Establishing

Version 2 - March 9, 2001

Simplified SIP Call Setup and

Bobs SIP Phone

200 (OK) F11

Version 2 - March 9, 2001

RTP MEDIA PATH

Simplified SIP Call Setup and

Simplified SIP Call Setup and

RTP MEDIA PATH

Version 2 - March 9, 2001

SIP Design Framework

Version 2 - March 9, 2001

Integration with IETF

Version 2 - March 9, 2001

Integration with IETF

Version 2 - March 9, 2001

Session Description Protocol

Ej. SDP Description

Version 2 - March 9, 2001

Version 2 - March 9, 2001

Version 2 - March 9, 2001

Version 2 - March 9, 2001

Feature Creation (2)

Version 2 - March 9, 2001

Version 2 - March 9, 2001

Describes terminals and other entities that

Version 2 - March 9, 2001

Call establishment and teardown.

Version 2 - March 9, 2001

Version 2 - March 9, 2001

H.323 Multipoint Control Unit

Version 2 - March 9, 2001

H.261 and H.263 Video codecs.

Call Control and

Call Control and Signaling

RAS - registration and other admission

Other ITU H. Recommendation

H.450.1 specifies framework for supplementary services. H.450.N

Specifies internetworking of H Series terminals with circuit switched

Version 2 - March 9, 2001

H.323 Components and

H.245 messages (optional)