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The Session Initiation Protocol (SIP) is a lightweight signaling protocol designed for initiating, managing, and terminating multimedia communication sessions. Standardized within the IETF MMUSIC working group, SIP facilitates both multicast and unicast communication, enabling personal mobility and flexible media changes. It supports features like call forwarding, resource reservation, and distinctive ringing. SIP operates over UDP or TCP and utilizes a text-based format similar to HTTP, making it suitable for deployment in intelligent networks. For detailed information, visit http://www.cs.columbia.edu/~hgs/sip.
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IETF Session Initiation Protocol Henning Schulzrinne Columbia University New York, NY
SIP Overview • Lightweight signaling protocol • to be standardized within IETF MMUSIC • invite to new and on-going conferences • multicast and unicast (“telephony”) • personal mobility • call center: reach first, first all • change and add media • post dial delay: 1.5 RTT • uses either UDP or TCP
SIP protocol • text based (~ HTTP) • methods
SIP for Intelligent Networks • IN services “inside” & “outside” network • UPN: translation by redirect server • call forwarding: proxy, end system • call transfer: BYE • distinctive ringing: end system • 800#: charging responsibility of resource reservation • call center (queueing, first available)
For more information... • http://www.cs.columbia.edu/~hgs/sip • IETF MMUSIC mailing list:confctrl-request@isi.edu