This is a repository for client-side WebRTC code samples and the AppRTC video chat client. The source for these samples is available at github.com/GoogleChrome/webrtc.
Some of the samples use new browser features. They may only work in Chrome Canary and/or Firefox Beta, and may require flags to be set.
Most of the samples use adapter.js, a shim to insulate apps from spec changes and prefix differences. (In fact, the standards and protocols used for WebRTC implementations are highly stable, and there are only a few prefixed names. For full interop information, see webrtc.org/interop.)
Please note that all samples that use getUserMedia() must be run from a server. Calling getUserMedia() from a file:// URL will result in a PERMISSION_DENIED NavigatorUserMediaError.
For more information about WebRTC, we maintain a list of WebRTC Resources. If you've never worked with WebRTC, we recommend you start with the 2013 Google I/O WebRTC presentation.
Patches and issues welcome!
Use getUserMedia with canvas and CSS filters
Audio-only getUserMedia() output to local audio element
Audio-only getUserMedia() displaying volume
Face tracking, using getUserMedia and canvas
Audio-only peer connection demo
Multiple peer connections at once
Forward the output of one PC into another
Use pranswer when setting up a peer connection
Display createOffer output for various scenarios
Display peer connection states
ICE candidate gathering from STUN/TURN servers
Web Audio output as input to peer connection
AppRTC video chat client powered by Google App Engine