SIP (Session Initiation Protocol)
Session Initiation Protocol was developed in the mid-1990s by the Internet Engineering Task Force as a real-time communication protocol for IP voice, and has expanded into video and instant-messaging applications.
In voice, the protocol performs basic call-control tasks such as session set up and tear down, or the signaling for call initiation, dial tone and termination. SIP also controls other signaling for features such as hold, caller ID and call transferring. Its functions are similar to the Signaling System 7 protocol in standard telephony and H.323 or Media Gateway Control Protocol in IP telephony.
According to SIP proponents, the protocol can provide converged and unified communication services, such as voice and video conversations. Like HTTP, SIP is a text-based protocol, which makes it easy to write applications that incorporate the technology, observers say.
The SIP model for telephony puts most of the intelligence for call setup and features on the SIP device or user agent - such as an IP phone or a PC with voice or instant-messaging software. That lets SIP user agents provide more features and operate in more of a peer-to-peer fashion. The method is different from traditional telephony or H.323-based telephony, where "dumb" phones are deployed, with most call processing and control intelligence residing on a centralized phone switch or server.
From IP telephony talk zeroes in on SIP, Network World, 04/15/02.
Also see SIMPLE, H.323 and session controller.
Additional resources
RFC 2543
Defines the protocol.
H.323 vs. SIP
Network World Convergence Newsletter, 07/03/02.
SIP breathes new life into voice over IP
Network World, 08/12/02.
Topic: Convergence
Latest news and analysis from Network World Fusion.
SIP Forum
Industry trade group.
Latest SIP news and analysis from Network World Fusion
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