Chapter 1. Introduction to SIP

The Session Initiation Protocol (SIP) was standardized by the Internet Engineering Task Force (IETF) and is described in several documents known as RFCs (Request for Comments). RFC3261 is one of the most recent and is called SIP version 2. SIP is an application-layer protocol used to establish, modify, and terminate sessions or multimedia calls. These sessions can be conferences, e-learning, telephony over the Internet, and similar applications. It is based on a text protocol similar to Hypertext Transfer Protocol (HTTP) and it is designed to start, keep, and close interactive communication sessions between users. These days SIP is one of the most used protocols for VoIP and is present on almost every IP phone in the market.

By the end of this chapter you will be able to:

Describe what SIP is
Describe what SIP is for
Describe SIP architecture
Explain the meaning of its main components
Understand and compare the main SIP messages
Describe the header fields processing for INVITE and REGISTER requests

The SIP protocol supports five features for establishing and closing multimedia sessions.

User location: Determines the endpoint address used for communication.
User parameters negotiation: Determines the media and parameters to be used.
User availability: Determines if the user is available or not to establish a session.
Call establishment: Establishes the parameters for caller and callee, and informs on call progress (ringing, ringback, congestion) to both parties.
Call management: Session transfer and closing.

The SIP protocol was designed as part of a multimedia architecture containing other protocols such as RVSP, RTP, RTSP, and SDP. However it does not depend on them to work.

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Building Telephony Systems with OpenSER

Chapter 1. Introduction to SIP