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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.
Marketing or business development professional who would like an introductory yet technical overview of the voice over IP protocols.
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Voice over IP Protocols
H. SIP supports TCP and UDP.323 H.323 and MGCP Call Control and Signaling H.SIP. Q. Version 2 .323 Version 1 and 2 supports H.225 H.931 over TCP and RAS over UDP.245 over TCP.323 Version 3 and 4 supports H.931 over UDP/TCP and RAS over UDP.931 RAS SIP MGCP RTP RTCP RTSP Signaling and Gateway Control Media Audio/ Video TCP IP UDP H. H.245 Q. 2001 4 .245 over UDP/TCP and Q.March 9.
Session Initiation Protocol .
An Session application layer signaling protocol that defines initiation.What is SIP? “ Initiation Protocol . IETF RFC 2543 Session Initiation Protocol ” 6 Version 2 . multimedia communication sessions between users. 2001 .March 9. modification and termination of interactive.
Version 2 .SIP Framework • Session initiation. 2001 7 . • Multiple users. • Interactive multimedia applications.March 9.
SIP Distributed Architecture SIP Components Location Server Redirect Server Registrar Server PSTN User Agent Proxy Server Proxy Server Gateway Version 2 .March 9. 2001 8 .
Version 2 . • User Agent Clients (UAC) – An entity that initiates a call. 2001 9 . • User Agent Server (UAS) – An entity that receives a call.User Agents An application that initiates.March 9. ØBoth UAC and UAS can terminate a call. receives and terminates calls.
2001 10 . rewrites or translates a request message before forwarding it. possibly after translation. • Interprets. to other servers.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. Version 2 .March 9.
March 9. 2001 11 .Location Server • A location server is used by a SIP redirect or proxy server to obtain information about a called party’s possible location(s). Version 2 .
the redirect server does not accept or terminate calls. • Unlike a proxy server.March 9.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 user agent server. the redirect server does not initiate its own SIP request. Version 2 . 2001 12 .
• The register server may support authentication. • A registrar server is typically co-located with a proxy or redirect server and may offer location services.March 9. 2001 13 . Version 2 .Registrar Server • A server that accepts REGISTER requests.
Successful Responses. • 5xx . • 4xx .Cancels a pending request. • OPTIONS – Used to query the capabilities of a server. • BYE . • 3xx . • REGISTER – Registers the user agent.March 9. . 2001 14 SIP Responses: • 1xx . • INFO – Used to carry out-of-bound information. • CANCEL . Version 2 .Server Failure Responses.Indicates termination of the call.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.Request Failure Responses.Redirection Responses.Global Failures Responses. • ACK . such as DTMF digits. • 2xx .Informational Messages. • 6xx .Confirms that the client has received a final response to an INVITE request.
168.180 SIP/126.96.36.199.168.36.21:5060 From: sip:email@example.com:5060 Content-Type: application/sdp Version 2 . • An example SIP header: ----------------------------------------------------------------SIP Header ----------------------------------------------------------------INVITE sip:firstname.lastname@example.org. 2001 15 . 1/ SIP enabled Accept: application/sdp Contact: sip:email@example.com/UDP 192.21 To: <sip:firstname.lastname@example.org Via: SIP/2.36. • A SIP messages looks like an HTTP message – message formatting.180> Call-ID: email@example.com.SIP Headers • SIP borrows much of the syntax and semantics from HTTP. header and MIME support.March 9.6.21 CSeq: 100 INVITE Expires: 180 User-Agent: Cisco IP Phone/ Rev.
1 – sip:14083831088@vovida.SIP Addressing • The SIP address is identified by a SIP URL. 2001 16 .10.168.org – sip:hostname@192. • Examples of SIP URLs: – sip:hostname@vovida. in the format: user@host.March 9.org Version 2 .
Version 2 .Process for Establishing Communication Establishing communication using SIP usually occurs in six steps: 1. 4. 2. 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. 3. Registering. Call setup. Determine the media to use – involves delivering a description of the session that the user is invited to. 6. 5.March 9. initiating and locating the user. Call modification or handling – example. Call termination. 2001 17 . call transfer (optional).
• Registration can also occur when the SIP user client needs to inform the proxy/registration server of its location. 200 – OK.March 9. the client registers with the proxy/registration server. 18 .Registration • Each time a user turns on the SIP user client (SIP IP Phone. or other SIP device). • The registration information is periodically refreshed and each user client must re-register with the proxy/registration server. 2001 SIP Phone User REGISTER 200 Proxy/ Registration Server REGISTER 200 Location/ Redirect Server SIP Messages: REGISTER – Registers the address listed in the To header field. PC. Version 2 . • Typically the proxy/registration server will forward this information to be saved in the location/redirect server.
Simplified SIP Call Setup and Teardown User Agent INVITE Proxy Server Location/Redirect Server INVITE 302 (Moved Temporarily) ACK INVITE Proxy Server User Agent Call Setup INVITE 302 (Moved Temporarily) ACK 180 (Ringing) 200 (OK) ACK 180 (Ringing) 200 (OK) ACK RTP MEDIA PATH BYE 200 (OK) BYE 200 (OK) BYE 200 (OK) INVITE 180 (Ringing) 200 (OK) ACK Media Path Call Teardown Version 2 .March 9. 2001 19 .
Version 2 .March 9.SIP – Design Framework SIP was designed for: • Integration with existing IETF protocols. • Scalability and simplicity. • Mobility. • Easy feature and service creation. 2001 20 .
• SAP Session Advertisement Protocol . SIP can works with existing IETF protocols. 2001 21 . • RTP Real Time Protocol -to transport real time data and provide QOS feedback.for controlling delivery of streaming media.March 9.for advertising multimedia session via multicast.Integration with IETF Protocols (1) Other IETF protocol standards can be used to build a SIP based application. • RTSP Real Time Streaming Protocol .to reserve network resources. Version 2 . for example: • RSVP .
Version 2 . • OSP – Open Settlement Protocol. 2001 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. • COPS – Common Open Policy Service.March 9. • HTTP – Hypertext Transfer Protocol .HTTP is the standard protocol used for serving web pages over the Internet.
Scalability • The SIP architecture is scalable. 2001 23 . – Functionality such as proxying. or registration can reside in different physical servers.March 9. redirection. Version 2 . flexible and distributed. location. – Distributed functionality allows new processes to be added without affecting other components.
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.
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• 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.
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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).
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• Presence. • Click to talk. • Unified messaging. • Find me / Follow me.March 9. Version 2 . • Instant messaging.). • Call forking.Feature Creation (2) SIP can support these features and applications: • Basic call features (call waiting. call blocking etc. 2001 27 . call forwarding.
edu/~hgs/sip/ Version 2 .References For more information on SIP refer to: IETF • http://www.org/html.cs.columbia.html Henning Schulzrinne's SIP page • http://www.March 9.ietf. 2001 28 .charters/sipcharter.
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.March 9.323 entities may provide real-time audio. 2001 .323 Version 4 ” 30 Version 2 . ITU-T Recommendation H. video and/or data communications. H.
March 9.H. 2001 31 .323 defines: • Call establishment and teardown. Version 2 .323 Framework H. • Audio visual or multimedia conferencing.
2001 32 .March 9.323 Components Gatekeeper Multipoint Control Unit Packet Based Networks Terminal Gateway Circuit Switched Networks Version 2 .H.
323 Terminals H.225 call control signaling.245 control channel signaling. • RTP/RTCP protocols for media packets. 2001 33 . • H.323 terminals are client endpoints that must support: • H. ØVideo codecs support is optional. • Audio codecs. Version 2 .H.March 9.
IP network) and circuit switched network (example. H.March 9. PSTN network). a gateway can provide translation between entities in a packet switched network (example. • Gateways can also provide transmission formats translation.323 and non-H. communication procedures translation.323 endpoints translations or codec translation.323 Gateway A gateway provides translation: • For example. Version 2 . 2001 34 .H.
323 system. • Call management (optional). • Bandwidth management (optional). • Zone management. Version 2 .H. Gatekeepers are optional but if present in a H. all H.March 9.323 Gatekeepers Gatekeepers provide these functions: • Address translation. • Admission control. • Call control signaling (optional). 2001 35 . • Call authorization (optional).323 endpoints must register with the gatekeeper and receive permission before making a call. • Bandwidth control.
video and/or data streams.H. Version 2 .323 Multipoint Control Unit MCU provide support for conferences of three or more endpoints. • Multipoint Processor (MP) – receives and processes audio. 2001 36 . An MCU consist of: • Multipoint Controller (MC) – provides control functions.March 9.
711 G. 2001 RTP UDP TCP IP TCP UDP TCP 37 .729 – Audio codecs.Capabilities advertisement. Audio Codec G.931 .225 Q.323 is an “Umbrella” Specification Media H.H. Version 2 .263 RTCP T.711. and conference control.729 Video Codec H. T.38 H. H.261 H.call signaling and call setup.245 Call Control and Signaling H. H.723 G.38 – Fax.120 T.323 Media Data/Fax Call Control and Signaling Data/Fax T.931 RAS H. RTP/RTCP – Media. RAS .registration and other admission control with a gatekeeper.March 9.245 .225 H. media channel establishment. G. G.723.120 – Data conferencing. G.225 Q.263 – Video codecs.261 and H.
1 specifies framework for supplementary services. H.235 H.N Description Specifies security and encryption for H.N recommendation specifies supplementary services such as call transfer.March 9. 2001 38 . Specifies internetworking of H Series terminals with circuit switched terminals. call diversion.323 and H. call hold. H.450. call park. name identification. call waiting.323 Protocol H.450. call completion. call offer. Recommendation that work with H. message waiting indication.246 Version 2 . H.245 based terminals. and call intrusion.450.Other ITU H.
245 messages over call control channel H.225/Q.245 messages (optional) H.Call Control.H. 2001 39 .931 (optional) H.323 endpoint and a gatekeeper.225 .323 Components and Signaling H.Q.225/Q.225/RAS messages over RAS channel H.225/Q.245 – A protocol for capabilities advertisement. media channel establishment and conference control. admission and status protocol used for communicating between an H. H.RAS – Registration.931 messages over call signaling channel H.March 9.245 messages (optional) Gatekeeper PSTN Gateway Terminal H. .225/RAS messages over RAS channel H. .931 (optional) H.931 – A protocol for call control and call setup. Version 2 .
2001 40 . Call setup.March 9. Initial communication and capabilities exchange. Call termination.323 may occurs in five steps: 1.Process for Establishing Communication Establishing communication using H. Audio/video communication establishment. 3. Version 2 . 4. Call services. 5. 2.
Connect H. ACF 7. • Terminal A sends a SETUP message to Terminal B. • The gatekeeper provides information for Terminal A to contact Terminal B. terminal capabilities. ARQ 6. Call Proceeding 5.Simplified H.323 Call Setup • Both endpoints have previously registered with the gatekeeper. ARQ 2.245 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.245 messages to determine master slave. • Terminal B and A exchange H.323 recommendation for more call setup scenarios.Alerting 8. • Terminal B sends a Alerting and Connect message. 2001 open logical channels. • Terminal A initiate the call to the gatekeeper. 41 . Terminal A Gatekeeper Terminal B 1. (RAS messages are exchanged).March 9. ACF 3. SETUP 4. • Terminal B responds with a Call Proceeding message and also contacts the gatekeeper for permission. and Version 2 . Refer to the H.
323 Version H.html http://www.packetizer.Versions of H. 2001 42 .323 Version 2 January 1998 http://www. Refer to the specification.March 9. http://www.com/iptel/h323/whatsnew _v4.323 Version 4 November 2000 Version 2 .html http://www.html H.com/iptel/h323/whatsnew _v2.packetizer.packetizer.packetizer.323 Version 3 September 1999 H.com/iptel/h323/whatsnew _v3.323 Version 1 Date May 1996 Reference for key feature summary New release.com/iptel/h323/ H.
References For more information on H.org Version 2 .com/iptel/h323/ Open H.packetizer. 2001 43 .int/itudoc/itu-t/rec/index.323 • http://www.itu.openH323.March 9.html Packetizer • http://www.323 refer to: ITU-T • http://www.
323 .Comparing SIP and H.
March 9. call setup and teardown. call identification. Both SIP and H. SIP and H. • Capabilities exchange.Comparing SIP and H.323 provide: • Call control. call return.323 are similar. • Basic call features such as call waiting. call transfer. call forwarding. 2001 45 . Version 2 . or call park. call hold.323 Similarities Functionally.
Work is in progress to add this functionality to H. H.323 Strengths • H. data collaboration.Comparing SIP and H.March 9.323 multimedia conferencing can support applications such as whiteboarding. or video conferencing. 2001 46 . • SIP – Third party call control is currently only available in SIP. • SIP – Supports flexible and intuitive feature creation with SIP using SIP-CGI (SIP-Common Gateway Interface) and CPL (Call Processing Language).323 – Defines sophisticated multimedia conferencing. Version 2 .323.
Peer-to-Peer.org/ 47 SIP ITU. location.323 Information Standards Body Relationship Origins IETF. and MGCP. H. and registration servers. H.323 terminals. 2001 .SIP and H.March 9. H. Widespread. SIP is gaining interest.SIG.323 Gatekeeper. Borrows call signaling protocol from ISDN Q. SIP proxy.323. Peer-to-Peer. Internet based and web centric. visit: http://www. Intelligent user agents. Intelligent H. Interoperability IMTC sponsors interoperability events among SIP. Interoperability testing between various vendor’s products is ongoing at SIP bakeoffs.323 Telephony based.Table 1 . Borrows syntax and messages from HTTP.imtc. Client Core servers Current Deployment Version 2 . redirect. For more information.
OSP to implement or enforce quality of service. 2001 . Text based UTF-8 encoding. Control Channel Encoding Type Server Processing Quality of Service Stateless or stateful. The H323 specification recommends using RSVP for resource reservation.1 PER encoding.323 gatekeeper. H. SIP relies on other protocols such as RSVP. H.323 Supported by H. 48 Version 2 . Version 1 or 2 – Stateful.SIP and H. Bandwidth management/control and admission control is managed by the H.323. COPS. SIP does not provide as extensive capabilities exchange as H. Version 3 or 4 – Stateless or stateful. Binary ASN.245 protocol.323 Information Capabilities Exchange SIP SIP uses SDP protocol for capabilities exchange.245 provides structure for detailed and precise information on terminal capabilities.March 9.Table 2 .
User agent registers with a proxy server. Uses E. Authentication and Encryption H. Redirect or location servers provide routing information.March 9. Authentication . Version 2 . Endpoint Location and Call Routing Uses SIP URL for addressing.323 system.323 Registration .235 provides recommendations for authentication and encryption in H.SIP and H.323 Information Security SIP Registration .323 systems. endpoints register and request admission with the gatekeeper. H.164 or H323ID alias and a address mapping mechanism if gatekeepers are present in the H.Table 3 .User agent authentication uses HTTP digest or basic authentication. 2001 49 .The SIP RFC defines three methods of encryption for data privacy. Encryption . Gatekeeper provides routing information.If a gatekeeper is present.
call waiting. H. call transfer.120 specification. Service or Feature Creation Supports flexible and intuitive feature creation with SIP using SIP-CGI and CPL. 2001 50 H. and call park.323 Basic call features.March 9. . call forwarding.323 Information Features Conferencing SIP Basic call features. Basic conferencing without conference or floor control. or find me/follow me. caller identification. Version 2 .450.1 defines a framework for supplementary service creation. Comprehensive audiovisual conferencing support. unified messaging. Data conferencing or collaboration defined by T. Note: Basic call features include: call hold. Some example features include presence.Table 4 – SIP and H.
pdf Version 2 .Reference This section cites a document that provides a comprehensive comparison on H. “Comparison of H. Ismail.323 and SIP: Dalgic.March 9. 1999. 2001 51 .323 and SIP for IP Telephony Signaling” in Proc.edu/~hgs/papers/others/ Dalg9909_Comparison. Fang.cs. of Photonics East. Hanlin.columbia. SPIE. (Boston. http://www. Sept. Massachusetts).
MGCP Media Gateway Control Protocol .
A protocol for controlling telephony gateways from external call control elements called media gateway controllers or call agents.What is MGCP? “ Media Gateway Control Protocol .March 9. IETF RFC 2705 Media Gateway Control Protocol ” 53 Version 2 . 2001 .
• Sends notification to the call agent about endpoint events. • Sends and receives commands to/from the gateway. Gateway • Provides translations between circuit switched networks and packet switched networks. control and processing intelligence to the gateway. • Execute commands from the call agents.Components Call agent or media gateway controller • Provides call signaling. 2001 Call Agent or Media Gateway Controller (MGC) SIP H. Version 2 .323 Call Agent or Media Gateway Controller (MGC) MGCP MGCP Media Gateway (MG) Media Gateway (MG) 54 .March 9.
The call agent sends commands to both gateways to establish RTP/RTCP sessions.Simplified Call Flow • When Phone A goes offhook Gateway A sends a signal to the call agent. The digits are forwarded to the call agent. The call agent sends commands to Gateway B. 2001 . Gateway B rings phone B. The call agent determines how to route the call.March 9. Gateway A generates dial tone and collects the dialed digits. Call Agent Media Gateway Controller • • • • • • MGCP MGCP RTP/RTCP Gateway A Gateway B Analog Phone A Analog Phone B 55 Version 2 .
MGCP Commands Call Agent Commands: • EndpointConfiguration • NotificationRequest • CreateConnection • ModifyConnection • DeleteConnection • AuditEndpoint • AuditConnection Gateway Commands: • Notify • DeleteConnection • RestartInProgress Version 2 . 2001 56 .March 9.
– Assumes limited intelligence at the edge (endpoints) and intelligence at the core (call agent).323 which are peer-to-peer protocols.March 9.323. • Interoperates with SIP and H. 2001 57 . – Differs from SIP and H.Characteristics of MGCP MGCP: • A master/slave protocol. Version 2 . – Used between call agents and media gateways.
– A call agent accepts SIP or H.323 and SIP. – The call agent uses MGCP to control the media gateway. – The media gateway establishes media sessions with other H. –A gateway that handles media.323 Gateway H. ØMGCP protocol is used to control the gateway.323 • MGCP divides call setup/control and media establishment functions.323. an H.March 9.323 provide symmetrical or peer-to-peer call setup/control. SIP and H. 2001 58 . H. • MGCP does not replace SIP or H. In this example. SIP and H.MGCP. For example.323 Gateway MGCP Media RTP/RTCP Media Gateway Version 2 . • MGCP interoperates with H.323 Call Agent/ Media Gateway Controller H.323 or SIP endpoints.323 gateway is “decomposed” into: –A call agent that provides signaling.323 call setup requests.
A user picks up analog phone and dials a number.RTP/ RTCP H. 2. 3. 3.Example Comparison H. The two gateways exchange capabilities information.323 Gateway MGCP 1.March 9. routing information. The gateway notifies call agent of the phone (endpoint) event. 5.323 Gateway H. The gateway determines how to route the call. and issues a command to the gateways to establish RTP/RTCP session with other end. The terminating gateway rings the phone. 2001 RTP/ RTCP Gateway B Gateway A Analog Phone 59 Analog Phone . The two gateways establish RTP/RTCP session with each other. The Call agent determines capabilities. 2. 3 1 5. 2 4 1 Analog Phone Call Agent/ Media Gateway Controller Analog Phone Version 2 . 4.323 1. A user picks up analog phone and dials a number.
http://www.packetizer.What is Megaco? A protocol that is evolving from MGCP and developed jointly by ITU and IETF: • Megaco . For more information refer to: • IETF .ITU.org/html.com/iptel/h248/ Version 2 .March 9.248 or H.ietf.charters/megacocharter.IETF.html • Packetizer . • H.http://www.GCP . 2001 60 .
org/rfc/rfc2705.References For more information on MGCP refer to: IETF • http://www.txt?number=2705 Version 2 .ietf.March 9. 2001 61 .
capabilities exchange.323 will provide the call control functionality and MGCP can be used to manage media establishment in media gateways.March 9. SIP or H. call control. call teardown.323 are comparable protocols that provide call setup.323.Summary • SIP and H. MGCP can be used with SIP or H. Version 2 . and supplementary features. 2001 63 . In a VoIP system. • MGCP is a protocol for controlling media gateways from call agents.
Additional References .
• http://www.internettelephony. MGCP. and H323 protocols.protocols.com Internet Telephony • http://www. 2001 65 .General VoIP Reference Pulver – IP Telephony News • http://www.pdf Version 2 .com/voip/posvoip.com An overview poster of the SIP.pulver.March 9.
End of Module This is the end of the VoIP Protocol Overview training module.org.March 9.vovida. For additional training and documentation visit us at www. 2001 66 . Version 2 .
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