This is a repository for the WebRTC Javascript code samples. The source for these samples is available at github.com/Temasys/Google-WebRTC-Samples.
It is originally a fork of github.com/webrtc/samples, but updated to integrate the Temasys Plugin, and work on Internet Explorer and Safari.
Some of the samples use new browser features. They may only work in Chrome Canary, Firefox Beta, Microsoft Edge (available with Windows 10), and latest versions of the Temasys Plugin and may require flags to be set.
Most of the samples use AdapterJS, 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, and plugin.temasys.com.sg.)
In Chrome and Opera, all samples that use getUserMedia() must be run from a server. Calling getUserMedia() from a file:// URL will work in Firefox and the Temasys Plugin, but fail silently in Chrome and Opera.
Use getUserMedia and play the video in a canvas
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
Choose camera, microphone and speaker
Choose media source and audio output
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
Peer connection as input to Web Audio
AppRTC video chat client powered by Google App Engine