WebRTC Draft Published
Written by Alex Denham   
Monday, 23 February 2015

The Web Real-Time Communications Working Group of W3C, World Wide Web Consortium, has published a Working Draft of WebRTC 1.0: Real-time communication between browsers.


The draft defines a set of ECMAScript APIs in WebIDL that will allow media to be sent to and received from another browser or device implementing the appropriate set of real-time protocols. The specification is being developed alongside a protocol specification developed by the IETF RTCWEB group. There’s also an API specification giving details on how to access local media devices.

WebRTC It is based on browser source code from Google that was made open source in June 2011. It is growing in importance with Firefox, Chrome and Opera all supporting WebRTC in public releases. It’s also part of HTML 5. WebRTC provides the means for real-time communications between browsers between web pages on a peer-to-peer basis.

While it is best known for its sound and video elements, there is a data side to WebRTC too. At the recent WebRTC Conference and Expo in Paris, StreamRoot, Viblast, Pipe and Mera showed off data channel solution, including an app for Facebook that provides peer-to-peer file transfer.

The latest specification covers a number of aspects of video-conferencing in HTML, starting with the ability to connect to remote peers using NAT-traversal technologies such as ICE, STUN, and TURN. The specification also lays out the details of how to send locally-produced streams to remote peers and to receive streams from remote peers. The final addition to the specification covers sending arbitrary data directly to remote peers.



Complete HTML5.0 Definition Published



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Last Updated ( Monday, 23 February 2015 )