SIP Protocol Short Overview

SIP RFC : RFC 2543

Session Initiation Protocol (SIP) is a application layer control simple signalling protocol for VoIP implementations using the Redirect Mode.

SIP is a textual client-server base protocol and provides the necessary protocol mechanisms so that the end user systems and proxy servers can provide different services:

Call forwarding in several scenarios: no answer, busy , unconditional, address manipulations (as 700, 800 , 900- type calls).
Callee and calling number identification
Personal mobility
Caller and callee authentication
Invitations to multicast conference
Basic Automatic Call Distribution (ACD)
SIP addresses (URL) can be embedded in Web pages and therefore can be integrated as part of powerful implementations (Click to talk, for example).

SIP using simple protocol structure, provides the market with fast operation, flexibility, scalability and multiservice support.

SIP provides its own reliability mechanism. SIP creates, modifies and terminates sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. Members in a session can communicate using multicast or using a mesh of unicast relations, or a combination of these. SIP invitations used to create sessions carry session descriptions which allow participants to agree on a set of compatible media types. It supports user mobility by proxying and redirecting requests to the user's current location. Users can register their current location. SIP is not tied to any particular conference control protocol. It is designed to be independent of the lower-layer transport protocol and can be extended with additional capabilities.

SIP transparently supports name mapping and redirection services, allowing the implementation of ISDN and Intelligent Network telephony subscriber services. These facilities also enable personal mobility which is based on the use of a unique personal identity

SIP supports five facets of establishing and terminating multimedia communications:

User location
User capabilities
User availability
Call setup
Call handling.

SIP can also initiate multi-party calls using a multipoint control unit (MCU) or fully-meshed interconnection instead of multicast. Internet telephony gateways that connect Public Switched Telephone Network (PSTN) parties can also use SIP to set up calls between them.

SIP is designed as part of the overall IETF multimedia data and control architecture currently incorporating protocols such as RSVP, RTP RTSP, SAP and SDP. However, the functionality and operation of SIP does not depend on any of these protocols.

SIP can also be used in conjunction with other call setup and signalling protocols. In that mode, an end system uses SIP exchanges to determine the appropriate end system address and protocol from a given address that is protocol-independent. For example, SIP could be used to determine that the party can be reached using H.323 to find the H.245 gateway and user address and then use H.225.0 to establish the call.

SIP Operation

Sip works as follows:
Callers and callees are identified by SIP addresses. When making a SIP call, a caller first locates the appropriate server and then sends a SIP request. The most common SIP operation is the invitation. Instead of directly reaching the intended callee, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies. Users can register their location(s) with SIP servers.

SIP messages can be transmitted either over TCP or UDP
SIP messages are text based and use the ISO 10646 character set in UTF-8 encoding. Lines must be terminated with CRLF. Much of the message syntax and header field are similar to HTTP. Messages can be request messages or response messages.

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