The Session Initiation Protocol (SIP) is a signalling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with buddy lists, and IP Centrex services.
Over the last couple of years, the Voice over IP community has adopted SIP as its protocol of choice for signalling. SIP is an RFC standard (RFC 3261) from the Internet Engineering Task Force (IETF), the body responsible for administering and developing the mechanisms that comprise the Internet. SIP is still evolving and being extended as technology matures and SIP products are socialised in the marketplace.
SIP is a request-response protocol that closely resembles two other Internet protocols, HTTP and SMTP (the protocols that power the world wide web and email); consequently, SIP sits comfortably alongside Internet applications. Using SIP, telephony becomes another web application and integrates easily into other Internet services. SIP is a simple toolkit that service providers can use to build converged voice and multimedia services.
In order to provide telephony services there is a need for a number of different standards and protocols to come together - specifically to ensure transport (RTP), to authenticate users (RADIUS, DIAMETER), to provide directories (LDAP), to be able to guarantee voice quality (RSVP, YESSIR) and to inter-work with today's telephone network.
SIP was designed to solve only a few problems and to work with a broad spectrum of existing and future IP telephony protocols. To this end SIP provides four basic functions. SIP allows for the establishment of user location (i.e. translating from a user's name to their current network address). SIP provides for feature negotiation so that all of the participants in a session can agree on the features to be supported among them. SIP is a mechanism for call management - for example adding, dropping, or transferring participants. And finally SIP allows for changing features of a session while it is in progress. All of the other key functions are done with other protocols.
It is important to remember the two basic assumptions on which SIP was designed
3G and SIP
The 3GPP (Third Generation Partnership Project) is producing globally applicable Technical Specifications and Technical Reports for a third generation mobile system. The group is using IP technology end-to-end to deliver multimedia content to mobile handsets so IETF protocols are a must. The call control and signalling function will be fulfilled by SIP.
Users will be identified by SIP URLs and/or E.164 numbers, the numbering system of the telephone system. The bearer system (GPRS or mobile IP) will manage micro-mobility. This is the movement of the mobile user from one base station to another. Macro-mobility, the movement of the mobile user from one domain to another, will be handled by SIP. SIP will route signalling so that services are available from the originating or terminating network
3GPP has identified the Call State Control Function (CSCF) in the network. This is the equivalent of a SIP server. There will be three different kinds of CSCF:
As SIP is rolled out as part of the 3G initiative, millions of SIP calls will be made.