This is a repository for client-side WebRTC code samples and the AppRTC video chat client.
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.
All 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/web-apis/interop.
NB: all samples that use getUserMedia() must be run from a server. Calling getUserMedia() from a file:// URL will result in a PermissionDeniedError NavigatorUserMediaError.
webrtc.org/testing lists command line flags useful for development and testing with Chrome.
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! See CONTRIBUTING for instructions. All contributors must sign a contributor license agreement before code can be accepted. Please complete the agreement for an individual or a corporation as appropriate. The Developer's Guide for this repo has more information about code style, structure and validation.
getUserMedia + canvas + CSS Filters
getUserMedia with resolution constraints
getUserMedia with camera/mic selection
Audio-only getUserMedia output to local audio element
Audio-only getUserMedia displaying volume
Multiple peer connections at once
Forward output of one peer connection into another
Use pranswer when setting up a peer connection
Adjust constraints, view stats
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