This is a signaling protocol used for establishing sessions in an IP network. SIP will assist you in attributing advance telephony services across the Internet. Internet telephony is evolving from its use as an “inexpensive” (but contemptible quality) way to make international phone calls to a serious business telephony capability. SIP sessions can include voice communications, instant messaging, and multimedia applications.
SIP is justified as a control protocol for creating, modifying and terminating session’s with one or more participants. These sessions include Internet multimedia conferences, Internet (or any IP Network) telephone calls and multimedia distribution. Members in a session can communicate via multicast or via a mesh of uncast relations, or via a combination of these.
SIP anticipates on a peer to peer architecture that uses intelligent network elements for advanced call processing and call management functions. These endpoints are accredited to as the user agent client and the user agent server. A proxy server can be used as an intermediary responsible for exchanging the request from the client to the SIP server. SIP proxy servers can provide advanced call processing functions including security, authentication, and call routing. Real-time Transport protocol (RTP) is used to carry the voice or video content at the application layer between SIP endpoints.
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