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Building Telephony Systems with OpenSIPS Second Edition - Second Edition
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Before we dive into OpenSIPS, it is very important to understand some important concepts related to Session Initiation Protocol (SIP). In this chapter, we will cover a brief tutorial regarding the concepts used later in this book. By the end of this chapter, we will have covered the following topics:
SIP was standardized by Internet Engineering Task Force (IETF) and is described in several documents known as Request for Comments (RFC). The RFC 3261 describes SIP version 2. SIP is an application layer protocol used to establish, modify, and terminate sessions or multimedia calls. These sessions can be audio and video sessions, e-learning, chatting, or screen sharing sessions. It is similar to Hypertext Transfer Protocol (HTTP) and designed to start, keep, and close interactive communication sessions between users. Nowadays, SIP is the most popular protocol used in Internet Telephony Service Providers (ITSPs), IP PBXs, and voice applications.
The SIP protocol supports five features to establish and close multimedia sessions:
The SIP protocol was designed as a part of a multimedia architecture containing other protocols such as Resource Reservation Protocol (RSVP), Real-Time Protocol (RTP), Real-Time Session Protocol (RTSP), Session Description Protocol (SDP), and Session Announcement Protocol (SAP). However, it does not depend on them to work.
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