Today, we are making available WebRTC, an open technology for voice and video on the web. With WebRTC, we’d like to make the browser the home for innovation in real time communications.

Until now, real time communications required the use of proprietary signal processing technology that was mostly delivered through plug-ins and client downloads. With WebRTC, we are open sourcing the voice and video engine technologies from our acquisition of GIPS, giving developers access to state of the art signal processing technology, under a royalty free BSD style license. This will allow developers to create voice and video chat applications via simple HTML and JavaScript APIs.

In this effort, we’ll be working closely with other browser developers such as Mozilla and Opera, to implement this technology for use by the broader web community.  In addition, we’ve collectively engaged with the standards communities such as IETF and W3C working groups to define and implement a set of standards for real time communications.

We expect more innovations in the coming months by various community members and we will continue to develop key technologies and features that enable open, real time communications on the web. 

A developer preview of WebRTC including source code, specs and tools is available now at code.webrtc.org.

Update 6/1/2011: Added Opera as a supporter of WebRTC.
Update 6/6/2011: Updated project URL

Posted by Rian Liebenberg, Engineering Director and Jan Linden, Product Manager, Google