TUTORIAL 5 - Bridging the gap between SIP and The WWW for an ubiquitous Real Time Communications network
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Abstract
WebRTC is an emerging technology which provides state-of-the-art native RTC (Real Time Communications) to web browsers and will be a very important piece for telecommunications in the coming years. By adding a signaling protocol for establishing media sessions, a browser becomes a full RTC communication device making the WWW the biggest RTC network that have ever existed.
Session Initiation Protocol (SIP) is a mature and widely used signaling protocol for real time session management. A recent IETF specification enables the use of SIP in the WWW by using WebSocket as a signaling transport: "The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP)". This new SIP specification facilitates the communication between web browsers and existing SIP devices and makes possible to accommodate web browses into existing SIP infrastructures.
Tutorial sponsored by ELASTIX.