webrtc-samples
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<h1>WebRTC samples</h1>
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<p>This is a repository for the WebRTC Javascript code samples. The source for these samples is available at <a href="https://github.com/webrtc/samples" title="View GitHub repository for these files">github.com/webrtc/samples</a>.</p>
<p>Some of the samples use new browser features. They may only work in <a href="https://www.google.com/chrome/browser/canary.html" title="Download Chrome Canary">Chrome Canary</a>, <a href="http://www.mozilla.org/firefox/beta/" title="Download Firefox Beta">Firefox Beta</a> or Microsoft Edge (available with Windows 10), and may require flags to be set.</p>
<p>Most of the samples use <a href="https://github.com/webrtc/adapter">adapter.js</a>, 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 <a href="http://www.webrtc.org/web-apis/interop">webrtc.org/web-apis/interop</a>.)</p>
<p>In Chrome and Opera, all samples that use <code>getUserMedia()</code> must be run from a server. Calling <code>getUserMedia()</code> from a file:// URL will work in Firefox, but fail silently in Chrome and Opera.</p>
<p><a href="http://www.webrtc.org/testing" title="Command-line flags for WebRTC testing">webrtc.org/testing</a> lists command line flags useful for development and testing with Chrome.</p>
<p>For more information about WebRTC, we maintain a list of resources at <a href="https://g.co/webrtc">g.co/webrtc</a>. If you've never worked with WebRTC, we recommend you start with the 2013 Google I/O <a href="https://www.youtube.com/watch?v=p2HzZkd2A40">WebRTC presentation</a>.</p>
<p>Patches and issues welcome! See <a href="https://github.com/webrtc/samples/blob/gh-pages/CONTRIBUTING.md">CONTRIBUTING.md</a> for instructions. The <a href="https://bit.ly/webrtcdevguide">Developer's Guide</a> for this repo has more information about code style, structure and validation. Head over to <a href="test/README.md">test/README.md</a> and get started developing.</p>
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<h2 id="the-demos">The demos</h2>
<h3 id="getusermedia">getUserMedia</h3>
<p><a href="src/content/getusermedia/gum/">Basic getUserMedia demo</a></p>
<p><a href="src/content/getusermedia/canvas/">Use getUserMedia with canvas</a></p>
<p><a href="src/content/getusermedia/filter/">Use getUserMedia with canvas and CSS filters</a></p>
<p><a href="src/content/getusermedia/resolution/">Choose camera resolution</a></p>
<p><a href="src/content/getusermedia/audio/">Audio-only getUserMedia() output to local audio element</a></p>
<p><a href="src/content/getusermedia/volume/">Audio-only getUserMedia() displaying volume</a></p>
<p><a href="src/content/getusermedia/face/">Face tracking, using getUserMedia and canvas</a></p>
<p><a href="src/content/getusermedia/record/">Record stream</a></p>
<h3 id="devices">Devices</h3>
<p><a href="src/content/devices/input-output/">Choose camera, microphone and speaker</a></p>
<p><a href="src/content/devices/multi/">Choose media source and audio output</a></p>
<h3 id="peerconnection">RTCPeerConnection</h3>
<p><a href="src/content/peerconnection/pc1/">Basic peer connection demo</a></p>
<p><a href="src/content/peerconnection/audio/">Audio-only peer connection demo</a></p>
<p><a href="src/content/peerconnection/multiple/">Multiple peer connections at once</a></p>
<p><a href="src/content/peerconnection/multiple-relay/">Forward the output of one PC into another</a></p>
<p><a href="src/content/peerconnection/munge-sdp/">Munge SDP parameters</a></p>
<p><a href="src/content/peerconnection/pr-answer/">Use pranswer when setting up a peer connection</a></p>
<p><a href="src/content/peerconnection/constraints/">Constraints and stats</a></p>
<p><a href="src/content/peerconnection/create-offer/">Display createOffer output for various scenarios</a></p>
<p><a href="src/content/peerconnection/dtmf/">Use RTCDTMFSender</a></p>
<p><a href="src/content/peerconnection/states/">Display peer connection states</a></p>
<p><a href="src/content/peerconnection/trickle-ice/">ICE candidate gathering from STUN/TURN servers</a></p>
<p><a href="src/content/peerconnection/restart-ice/">Do an ICE restart</a></p>
<p><a href="src/content/peerconnection/webaudio-input/">Web Audio output as input to peer connection</a></p>
<p><a href="src/content/peerconnection/webaudio-output/">Peer connection as input to Web Audio</a></p>
<h3 id="datachannel">RTCDataChannel</h3>
<p><a href="src/content/datachannel/basic/">Transmit text</a></p>
<p><a href="src/content/datachannel/filetransfer/">Transfer a file</a></p>
<p><a href="src/content/datachannel/datatransfer/">Transfer data</a></p>
<h3 id="videoChat">Video chat</h3>
<p><a href="https://apprtc.appspot.com/">AppRTC video chat client</a> powered by Google App Engine</p>
<p><a href="https://apprtc.appspot.com/params.html">AppRTC URL parameters</a></p>
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<a href="https://github.com/webrtc/samples" title="View the repository" id="viewSource">github.com/webrtc/samples</a>
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