SIP - Session Initiation Protocol | Voipproviderslist useful links
What is Session Initiation Protocol (SIP)The nuts and bolts
The Session Initiation Protocol (SIP) is a signal protocol employed to set up sessions in an IP network. A session could be a simple two-way voice call or it could be a conference session with shared multi-media. The capacity to establish these sessions means that a multitude of ground-breaking services turn out to be doable, for instance voice-enriched e-commerce, click-to-dial, Instant Messaging with buddy lists, etc.
Over the last two years, the Voice over IP folks have accepted SIP as the signal protocol of choice. SIP is an RFC standard (RFC 3261) from the Internet Engineering Task Force (IETF), which is the body responsible for administering and development of the tools that make the Internet. Needless to say that SIP is still new and evolving; it is being extended as technology matures.
The IETF's philosophy is one of minimalism: you only specify what you need to specify. SIP has been developed as a mechanism to establish sessions, it does not know about details of a session, it just initiates, terminates and modifies sessions. This minimalism means that SIP is extensible, and it can work comfortably in different architectures and deployment scenarios.
SIP is a request-response protocol very similar to two other Internet protocols, such as HTTP and SMTP (the of the world wide web and email); thus SIP operates well with Internet applications. Using SIP, telephony becomes one more web application and integrates easily into other Internet services. SIP is a simple tool that service providers can use to build converged voice and multimedia services.
If there is a need to provide telephony there is a need for a lot of various standards and protocols to work together, i.e. to ensure transport, to authenticate users, to provide directories , to guarantee voice quality and to co operate with the present days telephone network. Here we will only cover SIP.
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