abel diego - telefonía ip
TRANSCRIPT
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1© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005 111© 2004, Cisco Systems, Inc. All rights reserved.
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2© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
May 3rd, 2005
VoIPSignaling Protocols
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3© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
ABEL DIEGO RODRABEL DIEGO RODRÍGUEZÍGUEZSYSTEMS ENGINEER MANAGER
Public Sector
Mexico
Who I am …
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4© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Agenda
1. Making the Rules for VoIP
2. Distributed Control
3. Centralized Control
4. Detailed Call Flows
5. Final Remarks
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5© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Making the Rules for VoIP 1
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6© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
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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7© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Defining the VoIP Protocols
H.323H.323
An ITU Recommendation that defines “Packet-based multimedia communications systems”. H.323 defines a distributed architecture for creating multimedia applications, including VoIP
SIPSIP
Defined as IETF RFC 2543. SIP defines a distributed architecture for creating multimedia applications, including VoIP
MGCPMGCP
Defined as IETF RFC 2705. MGCP defines a centralized architecture for creating multimedia applications, including VoIP
H.248H.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”
MegacoMegacoDefined as IETF RFC 2885. Megaco defines a centralized architecture
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8© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
VoIP Signaling Protocols
Distributed ControlDistributed Control
H.323H.323
SIPSIP
Centralized ControlCentralized Control
MGCPMGCP
MEGACO / H.248MEGACO / H.248
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9© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Distributed Control 2
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10© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
G
S
T
N
Terminal
Terminal
ServerServerServer
Distributed Arquitecture
Gateway
Gateway
Gateway
Gateway
Gateway
Gateway
IP
G
S
T
NH.323
H.323
SIP
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11© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Endpoint’s Requirement
Each endpoint (Terminal or Gateway) is required to perform the following function:s
Media GatewayConvertion of media (voice, video, etc) from one network to the other (ie: TDM to IP)
Signaling It exchanges information with other devices in both networks using some kind of protocol (i.e.: SS7 in TDM networks, SIP in IP networks, etc)
Call ControlControls the parts of the call state that pertain to connection control for its media channels. Manages the dial plan (who can call who… when)
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12© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Endpoint’s Requirement
Signaling It exchanges information with other devices in both networks using some kind of protocol (i.e.: SS7 in TDM networks, SIP in IP networks, etc)
Call ControlControls the parts of the call state that pertain to connection control for its media channels. Manages the dial plan (who can call who… when)
Each endpoint (Terminal or Gateway) is required to perform the following function:s
Media GatewayWHAT DO THEY DO
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13© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Endpoint’s Requirement
Signaling It exchanges information with other devices in both networks using some kind of protocol (i.e.: SS7 in TDM networks, SIP in IP networks, etc)
Each endpoint (Terminal or Gateway) is required to perform the following function:s
Media GatewayWHAT DO THEY DO
Call ControlHOW THEY DECIDE WHAT TO DO
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14© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Endpoint’s Requirement
Each endpoint (Terminal or Gateway) is required to perform the following function:s
Media GatewayWHAT DO THEY DO
Signaling HOW THEY EXCHANGE INFORMATION ON WHAT THEY WILL DO
Call ControlHOW THEY DECIDE WHAT TO DO
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15© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
H.323
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16© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
H.323 Background
ITU recommendation
v1 approved in 1996
v2 approved in January 1998
v3 approved in September of 1999
V4 approved in 2000
Defines multimedia applications over packet-based networks
Voice coding is a requirement
Leverages existing standards
Wide market acceptance
Facilitates interoperability between vendors
Cisco VoIP solutions are H.323 compliant
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17© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
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
SpeechTerminalSpeech
TerminalH.320
TerminalH.320
TerminalSpeech
TerminalSpeech
Terminal
H.323 Components
e
GK
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18© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
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/O Equipment
User Data Applications
T.120, etc.
SystemControl
UserInterface
Video I/O Equipment
H.225Layer
UDP
TCP
UDP
IP
ReceivePain
Delay(Sync)
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19© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
H.245 (TCPDynamic Port)Open Logical Channel
Open Logical Channel Acknowledge
Capabilities Exchange / MSD
Media (UDP)RTP StreamRTP StreamRTCP StreamRTCP Stream
RTP StreamRTP Stream
VV
H.323Endpoint A
H.225 (TCP Port 1720)Setup
Alerting / Connect
H.323Endpoint B
VV
H.323 Signaling
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20© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
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
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21© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
H.225/H.245 Call Signalling
RTP Voice Bearer Traffic
H.225
RAS H.225 RAS
Directed Call Signalling
H.225/H.245Call Signalling
RTP Voice Bearer Traffic
H.225
RAS H.225 RAS
GK Routed Signalling (GRS)
GK Signalling Models
Cisco GKs do Direct mode only Many 3rd Party GKs are capable of both modes Scalability
Directed model scales better GK-based applications often require GRS
billingVPN servicescall deflection/control
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23© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP
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24© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
The history of SIP
Session Initiation Protocol (SIP) is defined via RFC2543 on March
17, 1999. Work was initiated in the IETF’s MMUSIC working-group.
Additional “feature” drafts have been written to address issues which
concern SS7/ISUP handling, QoS, Alerting, DHCP, 3PCC, Firewalls
& NAT, etc…
IETF SIP-WG created in September, 1999
RFC2543bis (additions) created in April 2000.
Vendor interoperability testing done at the semi-annual SIP Bakeoff
(5th in August, 2000 - NYC, 6th in December in Silicon Valley)
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25© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP @ the IETF – A lot going on
RFC 2543
Bis-01
Bis-02
Bis-03
Bis-04
Bis-05
Bis-06
Bis-07
Bis-08
Bis-09
1999REFERSession Timer
PrivacyMessage-Waiting
Diversion3PCC
PresenceSIP-T
Tel: URLTRIPINFOT.38
SupportedCaller-Pref
PRACK / COMETEtc..
SIP-WG
SIMPLE-WG
SIPPING-WG
RFC 3261 JUNE 2002
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26© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP Basics—Architecture
SIP utilizes an Internet architecture, similar to the World Wide Web—
intelligent clients utilizing services within the network
Applications and SIP services can be distributed throughout the
network, with intelligence in the clients or in applications on the network
Additional services like presence, mobility, find-me/follow-me, etc. are
added into the network via basic SIP functionality or application servers
Application servers may use LDAP, XML, IMAP, TAPI/JTAPI, CGI, CPL,
etc. to provide the back-end services
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27© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Open/Standard Interface
Open/Standard Interface
Open Packet Telephony and SIP
Open Service Application Layer
Open Service Application Layer
Open Call Control LayerOpen Call Control Layer
Standards-Based Packet Infrastructure Layer
Standards-Based Packet Infrastructure Layer
Connection PlaneConnection Plane
Call Control PlaneCall Control PlaneProxy Servers• Quality Of Service (Q0S)• Security• Routing• Privacy Control
Services PlaneServices Plane
V
Voice Gateways•26xx/36xx/AS5300IP Phones/Endpoints•Telecaster 7960•Partner Phones
Redirect/Application Servers•IP Centrex•ASP•Residential
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28© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP Basics—Architecture
Legacy PBX
SIP User Agents (UA)
Application Services
SIPSIPSIPSIP
RTPRTP
SIPSIP
CAS or PRI
SIP Proxy, Registrar and Redirect Servers
Inte
llig
ent
Ser
vice
sIn
telli
gen
t S
ervi
ces
PSTN
E-MailLDAP
Oracle XMLCPL
3PCC
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29© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP Basics - Architectural Elements
Clients:Clients: User Agent Client (UAC) / User Agent Server (UAS)
Originate & Terminate SIP requests
Typically an endpoint will have both UAC & UAS, UAC for originating requests, and UAS for terminating requests
Servers:Servers:Proxy Server
Redirect Server
Registrar Server
Back-to-Back User Agent (B2BUA)
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30© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP Model
SIPSIP is based on an HTTP-like request/response transactiontransaction model
Each transaction consists of a requestrequest that invokes a particular METHOD,
or function, on the server and at least one responseresponse
A SIP SIP METHOD specifies the action that the requestor wants the server to
take
METHODS contains a number of header fields
Header fields are named attributes that provide additional information
about a message
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31© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP Methods
Consists of Requests and Responses
Requests
REGISTER: UA registers with Registrar ServerINVITE: request from a UAC to initiate a sessionACK: confirms receipt of a final response to INVITEBYE: sent by either side to end a callCANCEL: sent to end a call not yet connectedOPTIONS: sent to query capabilities outside of SDP
SIP Responses: 1xx - Informational Messages. 2xx - Successful Responses. 3xx - Redirection Responses. 4xx - Request Failure Responses. 5xx - Server Failure Responses. 6xx - Global Failures Responses.
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32© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP Methods
Newly Adopted Methods:
SUBSCRIBE & NOTIFY: used to identify device status / presence.
The foundation of SIP IM / Presence (IMPP).
INFO: a means of carrying “data” in a message body
REFER: the mechanism to initiate a Transfer
MESSAGE: the means of carrying “data” for SIP IMPP
Messages contain SIP Headers and Body
Body might be SDP or an attachment or some other application
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33© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP Headers
-----------------------------------------------------------------
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
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35© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP - Register
SIP REGISTERREGISTER is sent by a non-Gateway UA (ie. SIP Phone) to the local SIP Server (Proxy/Redirect). The Server will give the REGISTER to the Registrar function. In this example, the SIP Phone is registering each DN line.
REGISTER sip:172.18.192.232 SIP/2.0Via: SIP/2.0/UDP 10.80.17.134:5060From: sip:[email protected]: sip:[email protected]: [email protected]: 100 REGISTERContact: <sip:[email protected]:5060>Expires: 3600Content-Length: 0
REGISTER sip:172.18.192.232 SIP/2.0Via: SIP/2.0/UDP 10.80.17.134:5060From: sip:[email protected]: sip:[email protected]: [email protected]: 100 REGISTERContact: <sip:[email protected]:5060>Expires: 3600Content-Length: 0
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36© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP - INVITE
INVITE sip:[email protected]:5060 SIP/2.0Expires: 180Content-Type: application/sdpVia: SIP/2.0/UDP 172.18.192.232:5060;branch=1FV1xhfvxGJOK9rWcKdAKOAVia: SIP/2.0/UDP 10.80.17.134:5060To: <sip:[email protected]>From: sip:[email protected]: [email protected]: 100 INVITEContact: sip:[email protected]:5060Content-Length: 219User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabledAccept: application/sdpRecord-Route: <sip:[email protected]:5060;maddr=172.18.192.232> v=0o=CiscoSystemsSIP-IPPhone-UserAgent 17045 11864 IN IP4 10.80.17.134s=SIP Callc=IN IP4 10.80.17.134t=0 0m=audio 29118 RTP/AVP 0 101a=rtpmap:0 pcmu/8000a=rtpmap:101 telephone-event/8000
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39© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Simplified SIP Call Flow
200 (OK)200 (OK)
180 (Ringing)180 (Ringing)
ACKACK
RTP MEDIA PATH
BYEBYE RELEASE
200 (OK)200 (OK)Call Teardown
MediaPath
Call Setup
INVITE
GatewayUser Agent
INVITE
Proxy Server
SETUP
PROCEEDING
ANSWER
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46© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
SIP - How the pieces fit in
User Agent
1 - INVITE
2 - INVITE
3 - 3XX Redirect
4 - INVITE
5 - 200 OKAY
6 - 200 OKAY
RTPV
SIP Proxy ServerSIP Redirect Server
SIP User AgentSIP User Agent
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47© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Centralized Control 3
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48© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Centralized Call Control
484848
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49© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Distributed Gateway Model
GatewayGateway GatewayGatewayIPIP
H.323
SIP
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50© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Distributed Gateway Model
GatewayGatewayIPIP
H.323
SIP
GatewayGateway
GatewayGateway
MGCMGC
MGMGSGSG
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51© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Distributed Gateway Model
Media Gateway Media Gateway ControllerController
Signaling Signaling GatewayGateway
Media GatewayMedia Gateway
IPIP
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52© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Distributed Gateway Model
Media Gateway Media Gateway ControllerController
Signalling Signalling GatewayGateway
Media GatewayMedia Gateway
Media GatewayMedia Gateway
The media gateway converts media provided in one type of network to the format required in another type of network; the gateway shall be capable of full duplex audio translations.
Media GatewayMedia Gateway
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53© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Distributed Gateway Model
Media Gateway Media Gateway ControllerController
Signalling Signalling GatewayGateway
Media GatewayMedia Gateway
Media Gateway Media Gateway ControllerController
Media Gateway ControllerMedia Gateway ControllerControls the parts of the call state that pertain to connection control for media channels in one or more media gateways (also referred to as a call agent, or the industry term “Softswitch”)
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54© 2005 Cisco Systems, Inc. All rights reserved. Cisco PublicVOIP PROTOCOLS OKON.JUL.2005
Distributed Gateway Model
Media Gateway Media Gateway ControllerController
Signalling Signalling GatewayGateway
Media GatewayMedia Gateway
Signaling GatewaySignaling GatewayThe Signaling Gateway is in the signaling (NOT media) path between SS7/ISDN signaling element, and call agent. Transforms SS7/ISDN signaling for transport on IP network. Signaling backhaul to media gateway controller.
Signaling Signaling GatewayGateway
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Distributed Gateway Model
Media Gateway Media Gateway ControllerController
Signaling Signaling GatewayGateway
Media GatewayMedia Gateway
sigtran megaco
mmusic
SCTP, IUA, M2UA, M2PA, M3UA, SUA,
V5UA
MGCP, MEGACO
SIP, SDP,SAP
Q.1901 (Q.BICC)
H.323
H.248
IETF stdsIETF stds
ITU stdsITU stds
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Distributed Gateway Model
Media Gateway Media Gateway ControllerController
Media GatewayMedia Gateway
sigtran megaco
mmusic
SCTP, IUA, M2UA, M2PA, M3UA, SUA
MGCP, MEGACO
SIP, SDP,SAP
Signaling Signaling GatewayGateway
E1/PRI
E1/R2
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Distributed Gateway Model
Media Gateway Media Gateway ControllerController
Media GatewayMedia Gateway
Signaling Signaling GatewayGateway
E1/PRI
E1/R2
Media GatewayMedia GatewaySignaling Signaling GatewayGateway
SS7
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Distributed Arquitecture
Media Gateway Media Gateway ControllerController
Signaling Signaling GatewayGateway
Media GatewayMedia Gateway
sigtran megaco
mmusicGateway
Gateway
Gateway
GatewayPSTN
IP
GATEWAY
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Distributed Arquitecture (ITU-T)
Media Gateway Media Gateway ControllerController
Signaling Signaling GatewayGateway
Media GatewayMedia Gateway
Gateway
Gateway
Gateway
GatewayPSTN
IP
H.323
GATEWAY
H.248H.248
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Distributed Arquitecture (IETF)
Media Gateway Media Gateway ControllerController
Signaling Signaling GatewayGateway
Media GatewayMedia Gateway
Gateway
Gateway
Gateway
GatewayPSTN
IPSIP
GATEWAY
MGCPMGCP
MEGACOMEGACO
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Distributed Arquitecture
Media Gateway Media Gateway ControllerController
Signaling Signaling GatewayGateway
Media GatewayMedia Gateway
Gateway
Gateway
Gateway
GatewayPSTN
IP
H.323
SIP
GATEWAY
MGCPMGCP
H.248H.248
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Recommendation H.248H.248 defines the protocols used between
elements of a physically decomposed multimedia gatewayphysically decomposed multimedia gateway,
used in accordance with the architecture as specified in
Recommendation H.323H.323.
There are no functional differences from a system view
between a decomposed gateway, with distributed sub-
components potentially on more than one physical device, and a
monolithic gateway such as described in Recommendation
H.246H.246.
H.248/MEGACOScope
Taken from the H.248 Recommendation (11/00)
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MGCP Concept
“MGCP is designed as an internal protocol within a distributed system that appears to the outside as a single VoIP gateway.”
RFC2705bis
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MGCP
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MGCP Overview
Media Gateway Control Protocol
IETF draft<draft-huitema-MGCP-v0r1-01.txt>
Dumb endpoint architecture
Centralized control architecture
Built as an extension to SGCP (Simple Gateway Control Protocol)
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History
SGCP (Simple Gateway Control Protocol)Designed by Bellcore and Cisco.
IPDC (Internet Protocol for Device Control)Design by Level 3 in conjunction with many vendors including Cisco.
MGCPMGCP is the result of merging SGCP and IPDC.
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MGCP versions
First draft dated Oct 11, 1998
A relatively stable version 0.1 dated Nov 9, 1998 <draft-huitema-MGCP-v0r1-01.txt>
MGCP 1.0 arrived at information RFC status on October, 1999 (RFC 2705)
MGCP 1.0 bis arrived at information RFC status on January, 2003 (RFC 3435)
MGCP is being extended with new packages:
RFC 3064: CAS PACKAGE
RFC 3149: Business Phone PACKAGE
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High Level Concept
Media Gateway (MG) contains “simple” endpoints
Services provided by “intelligent” Media Gateway Controller (MGC) or Call Agent (CA)
Endpoint provides user interactions, CA provides “centralized” call intelligence
Master/Slave relationship between CA and MG
“Simple” endpoint executes small set of simple transactions as instructed by CA
New services introduced by CA
Inexpensive endpoint can be mass-produced
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H.248 / MEGACO
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MEGACO Benefits
MEGACO is an adopted IETF/ITU standard
Cleaner connection model
Same things can be done in MGCP, but MEGACO has cleaner model
Better multimedia support
Theoretically yes via support of multiple streams, but more definition is required:
Has video support been defined?
Synchronization of voice/video/data been defined?
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Why should I deploy MGCP now if MEGACO is the official standard?
MGCP functionally equivalent to MEGACOMEGACO
MGCP is more mature protocol and is already in service provider betas
Interoperability and deployment issues have yet to be resolved in MEGACOMEGACO
Limited choice of call agents vendors with MEGACOMEGACO implementation
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Voice Myths
Networks can only be built one way
Networks will converge to IP
Myths Facts
Networks will only use one protocol
All networks will converge
H.323, SIP, MGCP and H.248/Megaco will all be present in VoIP networks
VoIP allows centralized or distributed architectures
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MGCP versus Megaco
Contrary to Popular Belief Megaco is not going to replace MGCP:
MGCP (NCS and TGCP) are PacketCable Specs
MGCP in the form of NCS and TGCP profiles are ITU-T recommendations (J.162 and J.171)
MGCP has continued to develop:
2705bis and basic packages are in the IETF editor’s queue (to be informational RFC’s); 2705bis allows significant extensibility mechanisms for the base protocol
IANA is in the process of setting up registration for MGCP packages
Additional packages: RFC 3064, bulk audits etc. continue to be developed
The Use of Megaco over MGCP or vice versa is strictly a marketing issue (there is no significant technical advantage of one over the other);
Example: MGCP limited to voice media (package extension all that’s required)
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Final Remarks 4
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• There are different VoIP protocols today, with different levels of maturity and different conceptual models
• H.323 and SIP are Distributed protocols (the intelligence is distributed)
• MGCP and MEGACO are Centralized protocols (the intelligence is centralized)
• VoIP networks, today, include a mix of these protocols and will increasingly do so in the future
VoIP Signaling ProtocolsConclusions
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