WebRTC code samples

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/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. See What are some chromium command-line flags relevant to WebRTC development/testing? for relevant flags.

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!

The demos

getUserMedia

Basic getUserMedia demo

getUserMedia + canvas

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

Face tracking

RTCPeerConnection

Basic peer connection

Audio-only peer connection

Multiple peer connections at once

Forward output of one peer connection into another

Munge SDP parameters

Use pranswer when setting up a peer connection

Adjust constraints, view stats

Display createOffer output

Use RTCDTMFSender

Display peer connection states

ICE candidate gathering from STUN/TURN servers

Web Audio output as input to peer connection

RTCDataChannel

Data channels

Video chat

Complete video chat client (based on Google App Engine)