SIP is based on RFC 2543  and is an application layer signaling protocol. It deals with interactive multimedia communication sessions between end-users, called user agents. It defines their initiation, modification, and termination. SIP calls may be terminal-to-terminal, or they may require a server to intercede. If a server is to be involved, it is only required to locate the called party. For interworking with non-IP networks, Megaco and H.323 are required. Often vendors of VoIP equipment integrate all three protocols on a single platform.
SIP is closely related to IP. SIP borrows most of its syntax and semantics from the familiar HTTP (hypertext transfer protocol). An SIP message looks very much like an HTTP message, especially with message formatting, header, and multipurpose Internet mail extension support. It uses addresses that are very similar to URLs (uniform resource locators) and to email. For example, a call may be made to [email protected] SIP messages are text-based rather than binary. This makes writing easier and the debugging of software more straightforward.
There are two modes with which a caller can set up a call with SIP. These are called redirect and proxy, and servers are designed to handle these modes. Both modes issue an "invite" message for another user to participate in a call. The redirect server is used to supply the address (URL) of an unknown called addressee. In this case the "invite" message is sent to the redirect server, which consults the location server for address information. Once this address information is sent to the calling user, a second "invite" message is issued, now with the correct address.
One specific type of SIP is called SIP-T (T for telephone). This is a function that allows calls from CCITT Signaling System 7 (SS7) to interface with a telephone in an IP-based network. The particular user part of SS7 for this application is ISUP (see Chapter 14, SS No. 7).
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