The animated TECHTip tutorial available here.

SIP [Session Initiation Protocol] is designed to perform basic call-control tasks, such as session call setup, tear down, and signaling for features such as call hold, caller ID, conferencing, and call transferring. However, with SIP, the intelligence for call setup and features resides on the SIP device or user agent, such as an IP phone, a PC with voice, or instant-messaging software. In contrast, traditional telephony or H.323-based telephony uses a model of intelligent, centralized phone switches with dumb phones with SS7 [Signaling System 7] in PSTN [Public Switched Telephone Network] telephone switching and H.323 or Media Gateway Control Protocol in IP telephony providing call control/routing. For more, go to sipforum.org.

SIP proxy server forks can be sequential or concurrent, and are useful in a call center or call hunting environment. Proxy server call flow is where the proxy server sets up the call. Proxy server is an optional SIP component that handles routing of SIP signaling but does not initiate SIP messages. Redirect server is an optional SIP component that does not route SIP messages. Redirect server returns a redirect (change in routing) to UA [User Agent] for direct routing (SIP is designed for end-to-end signaling without intervention by a server). The function of a presence server is to manage access, connections, directory (who’s who), billing and tracking, and other management functions. Note Internet access or VXML [Voice eXtensible Markup Language] applications can be directed based on an exception-triggered instant conference such as a disaster.

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