WebRTC code samples

This repository hosts forks of the different samples provided by Google with the few changes needed so they would work on IE and Safari with the plugin provided by Temasys

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.

NB for chrome: all samples that use getUserMedia() must be run from a server. Calling getUserMedia() from a file:// URL will result in a PERMISSION_DENIED NavigatorUserMediaError. See What are some chromium command-line flags relevant to WebRTC development/testing? for relevant flags.

Patches and issues welcome!

Tests updated today

getUserMedia()

getUserMedia with camera/mic selection

getUserMedia in iFrame

getUserMedia in iFrame

ICE candidate gathering

Peer Connection

Peer Connection States

Audio-only peer connection

Multiple peer connections

Multiple relays

Munge SDP

Data channels

Data channels with arraybuffers

AppRTC

Tests to be updated in the future

getUserMedia + canvas

getUserMedia + canvas + CSS Filters

getUserMedia with resolution constraints

Audio-only getUserMedia output to local audio element

Audio-only getUserMedia displaying volume

Face tracking

Accept incoming peer connection

Use pranswer when setting up a peer connection

Adjust constraints, view stats

Display createOffer output

Use RTCDTMFSender

Web Audio output as input to peer connection