SIP Communication

Session Initiation Protocol (SIP) is a signalling protocol used for establishing communication sessions within an IP based network. A session could be just as simple as a two-way telephone call or it could be a more advanced multi-media based conference session. The ability to establish these sessions allows additional innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with contact lists, and IP type centrex solutions.

In recent years Voice over IP innovators have adopted SIP as its preferred protocol of choice for data signalling. SIP communication complies to RFC standards (RFC 3261), established by the Internet Engineering Task Force (IETF), the body responsible for administering and developing the mechanisms that operate over the Internet. SIP is still an evolving technology which is being extended further as it starts to mature.

SIP communication is a request-response based protocol that closely resembles two other better know internet protocols, HTTP and SMTP (the protocols that power the world wide web and email); consequently, SIP sits comfortably alongside Internet based applications. Using SIP Communication, telephony services become another web type application and integrates easily into other internet services. SIP communication can be viewed as a simple toolkit that service providers can use to build converged voice and multimedia based services.

The IETF work to straightforward guidelines: specify only what you need to specify. SIP communication is very much of this mold; having been developed purely as a way to establish sessions, it does not know about the details of a session itself, it just initiates, terminates and modifies sessions. This simplicity means that SIP communication scales, it is extensible, and it sits very comfortably in different architectures and deployment scenarios.

When providing telephony type services using SIP communication there is a need for a number of different standards and protocols to be joined together - specifically to ensure transport (RTP), to authenticate users (RADIUS, DIAMETER), to provide directories (typically known as LDAP), to be able to guarantee voice quality (QoS, RSVP, YESSIR) and to interconect with today's "traditional" based telephone network (PSTN).

SIP Session Initiation Protocol (SIP Communication background)

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