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9. HTML5 WebRTC and SIP Over WebSockets

Presented by Thomas Quintana, TeleStax

HTML5 and WebRTC are free, open specifications that promise to enable rich, high quality, real-time communications applications to be developed in the browser via simple Javascript APIs and HTML5. Major browsers already support or will support it soon natively.

This talk will present an overview of WebRTC, why it can be highly disrupting for existing Operators and change the Telco industry. It will ask the question of where to place SIP when using WebRTC and describe SIP Over WebSockets, and how it can help operators bridge or expose their existing infrastructure to WebRTC, as well as what has to be done to accomplish it on a technical level and address concerns about Security, NAT Traversal, changes from RFC 3261, and other leveraged RFCs. It will also present the current challenges faced by both WebRTC and SIP Over WebSockets, and a live demo of a 1-to-1 WebRTC Video Conference will also be performed with a description of the call flow both at the signalling and media levels.