The document provides an overview of the Session Initiation Protocol (SIP), detailing its purpose in initiating, managing, and terminating communication sessions over the Internet. It outlines the architecture of SIP, including user agents and server types, as well as the various SIP messages and their functions. Additionally, it traces the history of SIP from its inception in 1996 to its standardization and use in cellular networks.
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(PDF) 3.3 - Session Initiation Protocol
The document provides an overview of the Session Initiation Protocol (SIP), detailing its purpose in initiating, managing, and terminating communication sessions over the Internet. It outlines the architecture of SIP, including user agents and server types, as well as the various SIP messages and their functions. Additionally, it traces the history of SIP from its inception in 1996 to its standardization and use in cellular networks.
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Signaling in Telecommunications
Session Initiation Protocol (SIP)
Outline
1. What are SIP used for ?
2. How does SIP work? 1. What are SIP used for ? • SIP is the core protocol for initiating, managing and terminating sessions in the Internet including text, voice, video or a combination of these. • SIP sessions can involve one or more participants and use unicast or multicast communications. • History of SIP – SIP was originally designed by university in 1996 – SIP was standardized as IETF RFC 2543 in 1999. – In November 2000, SIP was accepted as a 3GPP signaling protocol for IP-based streaming multimedia services in cellular networks. – In June 2002 the specification was revised in RFC 3261. 2. How does SIP work ? SIP architecture and entities • User Agent: entities involving in SIP – User Agent Client – User Agent Server
• Client: means end users e.g. softphone applications, messaging devices
• Server: – Proxy Server – Redirect server – Registrar server – Location server • A SIP address is written in a similar format to an email address e.g. sip:[email protected]
SIP messages: SIP is text-based protocol
• Requests – sent from client to server – INVITE :Invites a user to a call – ACK : Acknowledgement is used to facilitate reliable message exchange for INVITEs. – BYE :Terminates a connection between users – CANCEL :Terminates a request, or search, for a user. It is used if a client sends an INVITE and then changes its decision to call the recipient. – OPTIONS :Solicits information about a server's capabilities. – REGISTER :Registers a user's current location – INFO :Used for mid-session signaling • Responses – sent from server to the client which are categorized in series of codes denoted as 1xx, 2xx … 6xx – 1xx: Provisional -- request received, continuing to process the request; – 2xx: Success -- the action was successfully received, understood, and accepted; – 3xx: Redirection -- further action needs to be taken in order to complete the request; – 4xx: Client Error -- the request contains bad syntax or cannot be fulfilled at this server; – 5xx: Server Error -- the server failed to fulfill an apparently valid request; – 6xx: Global Failure -- the request cannot be fulfilled at any server. • INVITE sip:[email protected] SIP/2.0 (Message Headers) Via: SIP/2.0/UDP 10.10.10.10:5060 From: "Me" <sip:[email protected]>;tag=a0 To: "User" <sip:[email protected]> Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:10.10.10.10:5060> User-Agent: SIPTelephone Content-Type: application/sdp Content-Length: 251 (Message Body) v=0 o=audio1 0 0 IN IP4 10.10.10.10 s=session c=IN IP4 10.10.10.10 m=audio 54742 RTP/AVP 4 3 a=rtpmap:4 G729/8000 a=rtpmap:3 GSM/8000 Connecting between SIP and PSTN