TUTORIAL 5 - Bridging the gap between SIP and The WWW for an ubiquitous Real Time Communications network

on .


José Luis Millán Villegas, Palosanto Solutions


4 hours


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.

Biography of the lecturer

José Luis Millán Villegas is a Telecommunications Engineer from the University of the Basque Country in Bilbao, Spain. He started working in VoIP in late 2008 and works for a ITSP in Madrid, Spain, providing large scale VoIP solutions based on open standards mainly using the SIP (Session Initiation Protocol). He started his research for integrating the WWW and SIP in early 2011 and is a co-author of the IETF dratf "The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP)". He is the main author of JsSIP "The JavaScript SIP library", a fully capable SIP stack using WebSocket as a transport and WebRTC for integrating RTC (Real Time Communications) within WebRTC devices. His mission is unifying the WWW and the SIP world into a wider one.