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Session Initiation Protocol
(SIP) is a protocol developed by the IETF MMUSIC Working Group
and the proposed standard for initiating, modifying, and terminating
an interactive user session that involves multimedia elements
such as video, voice, instant messaging, online games, and virtual
reality. Initially it was published in 1996 as RFC 2543, now
obsolete, due to the publication of the new RFC
3261 in 2002
The main objective of SIP is the communication between
multimedia devices. SIP makes the communication possible thanks
to two protocols: RTP/RTCP and SDP.
RTP Protocol is used to transport voice data in real
time (the same as H.323
protocol), whereas SDP
protocol is used to negotiate the participant capabilities,
codification type, etc.
SIP has been designed in conformance with the Internet model.
It is an end-to-end oriented signaling protocol which means,
that all the logic is stored in end devices (except routing
of SIP messages). State is also stored in end-devices only,
there is no single point of failure and networks designed this
way scale well. The price that we have to pay for the distributiveness
and scalability is higher message overhead, caused by the messages
being sent end-to-end.
Therefore, SIP is an application-layer control protocol,
a Signaling protocol for Internet Telephony. SIP can establish
sessions for features such as audio/videoconferencing, interactive
gaming, and call forwarding to be deployed over IP networks
thus enabling service providers to integrate basic IP telephony
services with Web, e-mail, and chat services. It is based on
request and answer messages and reuses many concepts of previous
standards like HTTP and SMTP. |
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