SIP (Session Initiation Protocol)

An Application Layer signaling protocol for establishing, modify-

ing, and terminating multimedia sessions or calls over an IP network.The IETF defined SIP in RFC 2543

(1999), which was replaced by RFC 3261 (2002). SIP is a modular component of IP telephony, although it can function over any network. SIP offers considerable advantages over H.323, which is criticized for being overly complex and highly centralized. SIP was built specifically for an IP environment in which intelligence is highly decentralized in a large number of client agent servers. SIP identifies clients through a hierarchical URL similar to an e-mail address. A calling client can initiate a call in several ways. If the SIP address of the destination SIP client is known, the calling client simply sends the destination client an invite message, in care of a local proxy server.The proxy server sends the invite message to the distant proxy server, inviting the destination endpoint to join the session, and providing it with enough information to do so. If the SIP address of the destination SIP client is unknown, the calling proxy server sends the invite message to a redirect server, which consults the location server for address information.The redirect server passes that information to the calling proxy server, which then issues an invite message to the distant proxy server, including the information required to join the call. If the call is to a call center, such information might include a request to employ H.261 video, G.728 audio, and Spanish as the preferred language.The proxy server on the receiving end might consult an optional SIP location server on the receive end to determine the exact location of the called client and connect the call.This approach is much simpler and faster than the back-and-forth process involved in H.323, although layers of complexity are being added as the standards process works to enhance SIP to match H.323 and PSTN functionality.When the called client receives the invitation to join the session, it can either accept the call, or forward it to a messaging system or a user, perhaps a Spanish-speaking call center agent. Assuming that the call is a multimedia call comprising both video and voice, the called client (or messaging system) can elect either to accept the composite call or to accept only one of the datastreams, perhaps rejecting the video call but accepting the voice call. SIP also supports call forking, or splitting, so that several client extensions can be rung at once.

See also Application Layer, call, client, endpoint, G.728, H.261, IETF, IP, multimedia, protocol, proxy server, PSTN, server, session, and URL.

• sideband
• side circuits
• side circuits
• signal
• signal control point (SCP)
• signaling and control
• signaling rate
• signaling speed
• Signaling System 7 (SS7)
• signal-to-noise ratio (SNR)
• signal transfer point (STP)
• silence suppression
• silica
• silicon dioxide (SiO )
• silicone
• silicon (Si)
• Simple Gateway Control Protocol (SGCP)
• Simple Mail Transfer Protocol (SMTP)
• simple mode
• Simple Network Management Protocol (SNMP)
• SIMPLE (Session Initiation Protocol [SIP] for Instant Messaging and Presence Leveraging
• simplex
• simplex
• simplex
• simplified message desk interface (SMDI)