What is SIP?

Internet and mobile communication have taken over the individual as well as the business communication methods on a large scale. The use of instant messaging applications is increasing year after year. In this scenario Session Initiation Protocol (SIP) plays a very crucial role.

A close monitoring of the telecom world will show that it is slowly shifting from separate voice and data networks to a single converged network for all forms of communication. All multimedia applications are usable over a common IP-based hardware/software infrastructure including both fixed and mobile elements and broadband transmission. SIP will equip the platforms for this IP Multimedia Subsystem (IMS).

SIP is an application layer protocol, RFC standard (RFC 3261). It was developed and designed within the IETF (Internet Engineering Task Force) MMUSIC (Multiparty Multimedia Session Control) as an alternative to H.323. SIP is a request-response signaling protocol (system of formal rules) for setting up and starting voice, video, and instant messaging communication sessions over Internet.

SIP, however, is independent of the details of the session. Sessions here mean a set of senders and receivers that communicate and the state they are kept in during the communication. SIP isn't all in one; it just makes the communication possible - the communication itself is achieved by other means.

SIP is neither a session description protocol (SDP), nor does it provide conference control. The main advantage is that it is compatible with different architectures and deployment scenarios in the Internet services. Its essential communication function is aided by extensions and further protocols and standards. Two protocols commonly used are: RTP and SDP.

RTP (Real-time Transport Protocol) is used to carry the real time multimedia data like audio, video, and text. This protocol encodes and splits the data into packets and transports such packets over the Internet. SDP describes and encodes capabilities of session participants. Such a description is then used to negotiate the characteristics of the session so that all devices can participate.

SIP's main function is to help session originators deliver invitations to potential session participants irrespective of where they are. To achieve this SIP uses a wide variety of protocols. Each protocol distinctly addresses the different aspects of the requirement. Some of the protocols are: SOAP, HTTP, XML, VXML, WSDL, and SDP. SIP controls the functioning of MGCP and MEGACO also. Some specific functions are:

a) Name translation and user location - ensuring that the called party is located and the call reaches it.

b) Feature negotiation - coming to an agreement on the features to be supported by the group involved in the call.

c) Call participant management - the user can bring other users in to the call or put them on hold or cancel their call while the session is on.

d) Call feature changes - enabling the user to change call characteristics like voice-only facility for one user,video function for another during the same session.

e) SIP also enables other capabilities like encryption and security.