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Session Initiation Protocol (SIP)

SIP  
Session Initiation Protocol (SIP) was developed by the Internet Engineering Task Force (IETF) MMUSIC working group for the purpose of initiating, modifying, and terminating multimedia transmissions. Although SIP can be used as a catalyst for the transmission of such elements as video, instant messaging, and online games, SIP has become most well known for its predominance among VoIP signaling protocols for it's ability to efficiently initiate voice transmissions over peer-to-peer (P2P) connections or through IP communications networks.

Session Initiation Protocol (SIP) serves four major purposes - it establishes the location of the user, it establishes which features will be enabled during the session, it manages which components of the session are added or dropped, and it allows for the modification of features. All other functions of VoIP and other multimedia transmissions are handled by other protocols. The true power of SIP, however, is in it's ability to seamlessly integrate with these other protocols as well as protocols which overlap its own functions. SIP can seamlessly integrate and work with H.323, MGCP, and MEGACO telephony protocols as well as HTTP, XML, VXML, WSDL, SDP, SOAP, UDDI, and many others as well.

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