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SIP Makes Your Presence Known

SIP extensions let users see whether a colleague is online and available for VoIP calls, videoconferences and IM.

The SIP proxy server also can make intelligent routing decisions. If a destination UA such as a SIP phone isn't available and sends a BUSY HERE signal (486), the proxy server can send the INVITE to a designated voicemail server, ring multiple endpoints simultaneously or follow a route determined by a unified messaging server's Find me, Follow me algorithm. It also advertises whether a UA is available to participate in a multimedia session such as a VoIP call, online chat or IM session and identifies the media types the UA can support, such as audio and video.

Once a multimedia session is established, SIP transfers the session to another endpoint (using call forwarding, for example), adds other endpoints to an existing session or modifies the parameters to include other multimedia data. When an endpoint sends a BYE message, SIP terminates the session by hanging up a phone or closing an IM application (for more on how SIP works, see It's Time To Take a Look At SIP ).

The SIP header fields include named attributes identifying the parties in a session, plus information about the message format, using the IETF's SDP (Session Description Protocol). An INVITE request, for example, includes a unique identifier for the call, the destination address (URI) and the type of session a UA wants to establish, such as an application or protocol.

With SDP, the body of a SIP message can contain details about a multimedia data-exchange session and a UA's ability to support the requested exchange. Either RTP (Real-Time Transport) or RTSP (Real-Time Streaming Protocol) will work for streaming audio and video because SIP operates independently of these protocols as well as the underlying transport.

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