WebRTC From Wikipedia, the free encyclopedia Web
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WebRTC
WebRTC ( Web Real-Time Communication ) is an API definition being drafted by the World Wide Web Consortium (W3C) to enable browser -to-browser applications for voice calling , video chat , and P2P file sharing without plugins .
History
A project known as WebRTC, for browser-based real-time communication, was open sourced by Google in May 2011. [ 1 ] This has been followed by ongoing work to standardise the relevant protocols in the IETF [ 2 ] and browser APIs in the W3C. [ 3 ]
The W3C draft of WebRTC [ 4 ] is a work in progress with advanced implementations in the Chrome and Firefox browsers. The API is based on preliminary work done in the WHATWG . [ 5 ] It was referred as the ConnectionPeer API, and a pre-standards concept implementation was created at Ericsson Labs . [ 6 ] The Web Real-Time Communications Working Group expects this specification to evolve significantly based on:
- The outcomes of ongoing exchanges in the companion RTCWEB group at IETF [ 7 ] to define the set of protocols that, together with this document, will enable real-time communications in Web browsers.
- Privacy issues that arise when exposing local capabilities and local streams.
- Technical discussions within the group, on implementing data channels in particular. [ 8 ]
- Experience gained through early experimentation.
- Feedback received from other groups and individuals.