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01 June 2011, 12:45

Google open source WebRTC for open video/audio chat

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Web RTC logo Google has released a framework for "real time communications in the browser", called WebRTC, as open source. It is an implementation of current proposals which allow developers to created audio and video chat and conferencing applications which run within the web browser by using JavaScript APIs to control the network connections and audio/video encoding and decoding.

For example, a social networking web site could be extended to connect friends using WebRTC video chats or a collaborative development web site could create audio conference calls for project teams. The WebRTC initiative is supported by Google and Mozilla.

Zoom The architecture of WebRTC
Google's WebRTC codecs and intellectual property are all royalty-free, come with a patent grant and the source code is available under a BSD licence. Google obtained the underlying audio technology when it acquired Global IP Solutions in May 2010. GIPS was known for it's narrowband iLBC and wideband iSAC voice codecs. These codecs, along with Google's VP8 video codec are brought together in a communications stack which includes network connection technology, in part from Google Talk's libjingle. The network components manage NAT and firewall traversal, implement buffering and error concealment for the audio and video streams and offer support for peer to peer connections.

Google Talk already uses iSAC as it's default voice codec and the company says it is in the early stages of working on using WebRTC for Google Talk and Android. According to the projects FAQ, the API is still in development but is working through the WHATWG, W3C and IETF to establish a stable API implemented in "a few browsers". Once stabilised it hopes then to maintain backwards compatibility and interoperability. WebRTC source code is avalable form the projects Google Code page.


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