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<h1>WebRTC samples</h1>
<section>
<p>
This is a collection of small samples demonstrating various parts of the <a
href="https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API">WebRTC APIs</a>. The code for all
samples are available in the <a href="https://github.com/webrtc/samples">GitHub repository</a>.
</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.</p>
<p><a href="http://www.webrtc.org/testing" title="Command-line flags for WebRTC testing">https://webrtc.org/getting-started/testing</a>
lists command line flags useful for development and testing with Chrome.</p>
<p>Patches and issues welcome! See <a href="https://github.com/webrtc/samples/blob/gh-pages/CONTRIBUTING.md">CONTRIBUTING.md</a>
for instructions.</p>
<p class="warning"><strong>Warning:</strong> It is highly recommended to use headphones when testing these
samples, as it will otherwise risk loud audio feedback on your system.</p>
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<section>
<h2 id="getusermedia"><a href="https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia">getUserMedia():</a>
</h2>
<p class="description">Access media devices</p>
<ul>
<li><a href="src/content/getusermedia/gum/">Basic getUserMedia demo</a></li>
<li><a href="src/content/getusermedia/canvas/">Use getUserMedia with canvas</a></li>
<li><a href="src/content/getusermedia/filter/">Use getUserMedia with canvas and CSS filters</a></li>
<li><a href="src/content/getusermedia/resolution/">Choose camera resolution</a></li>
<li><a href="src/content/getusermedia/audio/">Audio-only getUserMedia() output to local audio element</a>
</li>
<li><a href="src/content/getusermedia/volume/">Audio-only getUserMedia() displaying volume</a></li>
<li><a href="src/content/getusermedia/record/">Record stream</a></li>
<li><a href="src/content/getusermedia/getdisplaymedia/">Screensharing with getDisplayMedia</a></li>
<li><a href="src/content/getusermedia/pan-tilt-zoom/">Control camera pan, tilt, and zoom</a></li>
</ul>
<h2 id="devices">Devices:</h2>
<p class="description">Query media devices</p>
<ul>
<li><a href="src/content/devices/input-output/">Choose camera, microphone and speaker</a></li>
<li><a href="src/content/devices/multi/">Choose media source and audio output</a></li>
</ul>
<h2 id="capture">Stream capture:</h2>
<p class="description">Stream from canvas or video elements</p>
<ul>
<li><a href="src/content/capture/video-video/">Stream from a video element to a video element</a></li>
<li><a href="src/content/capture/video-pc/">Stream from a video element to a peer connection</a></li>
<li><a href="src/content/capture/canvas-video/">Stream from a canvas element to a video element</a></li>
<li><a href="src/content/capture/canvas-pc/">Stream from a canvas element to a peer connection</a></li>
<li><a href="src/content/capture/canvas-record/">Record a stream from a canvas element</a></li>
<li><a href="src/content/capture/video-contenthint/">Guiding video encoding with content hints</a></li>
</ul>
<h2 id="peerconnection"><a href="https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection">RTCPeerConnection:</a>
</h2>
<p class="description">Controlling peer connectivity</p>
<ul>
<li><a href="src/content/peerconnection/pc1/">Basic peer connection demo</a></li>
<li><a href="src/content/peerconnection/perfect-negotiation/">Peer connection using Perfect Negotiation</a></li>
<li><a href="src/content/peerconnection/audio/">Audio-only peer connection demo</a></li>
<li><a href="src/content/peerconnection/bandwidth/">Change bandwidth on the fly</a></li>
<li><a href="src/content/peerconnection/change-codecs/">Change codecs before the call</a></li>
<li><a href="src/content/peerconnection/upgrade/">Upgrade a call and turn video on</a></li>
<li><a href="src/content/peerconnection/multiple/">Multiple peer connections at once</a></li>
<li><a href="src/content/peerconnection/multiple-relay/">Forward the output of one PC into another</a></li>
<li><a href="src/content/peerconnection/munge-sdp/">Munge SDP parameters</a></li>
<li><a href="src/content/peerconnection/pr-answer/">Use pranswer when setting up a peer connection</a></li>
<li><a href="src/content/peerconnection/constraints/">Constraints and stats</a></li>
<li><a href="src/content/peerconnection/old-new-stats/">More constraints and stats</a></li>
<li><a href="src/content/peerconnection/create-offer/">Display createOffer output for various scenarios</a>
</li>
<li><a href="src/content/peerconnection/dtmf/">Use RTCDTMFSender</a></li>
<li><a href="src/content/peerconnection/states/">Display peer connection states</a></li>
<li><a href="src/content/peerconnection/trickle-ice/">ICE candidate gathering from STUN/TURN servers</a>
</li>
<li><a href="src/content/peerconnection/restart-ice/">Do an ICE restart</a></li>
<li><a href="src/content/peerconnection/webaudio-input/">Web Audio output as input to peer connection</a>
</li>
<li><a href="src/content/peerconnection/webaudio-output/">Peer connection as input to Web Audio</a></li>
<li><a href="src/content/peerconnection/negotiate-timing/">Measure how long renegotation takes</a></li>
</ul>
<h2 id="datachannel"><a
href="https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel">RTCDataChannel:</a></h2>
<p class="description">Send arbitrary data over peer connections</p>
<ul>
<li><a href="src/content/datachannel/basic/">Transmit text</a></li>
<li><a href="src/content/datachannel/filetransfer/">Transfer a file</a></li>
<li><a href="src/content/datachannel/datatransfer/">Transfer data</a></li>
<li><a href="src/content/datachannel/messaging/">Messaging</a></li>
</ul>
<h2 id="videoChat">Video chat:</h2>
<p class="description">Full featured WebRTC application</p>
<ul>
<li><a href="https://apprtc.appspot.com/">AppRTC video chat client</a> powered by Google App Engine</li>
<li><a href="https://apprtc.appspot.com/params.html">AppRTC URL parameters</a></li>
</ul>
<h2 id="capture">Insertable Streams:</h2>
<p class="description">API for processing media</p>
<ul>
<li><a href="src/content/insertable-streams/endtoend-encryption">End to end encryption using WebRTC Insertable Streams</a></li> (Experimental)
<li><a href="src/content/insertable-streams/video-analyzer">Video analyzer using WebRTC Insertable Streams</a></li> (Experimental)
<li><a href="src/content/insertable-streams/video-processing">Video processing using MediaStream Insertable Streams</a></li> (Experimental)
<li><a href="src/content/insertable-streams/audio-processing">Audio processing using MediaStream Insertable Streams</a></li> (Experimental)
</ul>
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