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<h2>World's first HTML5 SIP client</h2>
<p>This is the world's first open source (<a target=_blank href="http://opensource.org/licenses/BSD-3-Clause">BSD license</a>) HTML5 SIP client entirely written in javascript for integration in social networks (FaceBook, Twitter, Google+), online games, e-commerce websites, email signatures... No extension, plugin or gateway is needed. The media stack rely on <a href="http://en.wikipedia.org/wiki/WebRTC" target="_blank">WebRTC</a>.<br />
The client can be used to connect to any SIP or IMS network from your preferred browser to make and receive audio/video calls and instant messages.<br />
</p>
<p><a class="btn btn-primary btn-large" href="call.htm?svn=203" target="_blank">Enjoy our live demo »</a></p>
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<tr><td><h2><a name="aSipSdpStack">Javascript SIP/SDP stack</a></h2></td></tr>
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<p>The SIP and SDP stacks (<b>~1 Mo</b>) are entirely written in javascript and the network transport uses WebSockets as per <a href="http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket">draft-ibc-sipcore-sip-websocket</a>.
The <a href="call.htm?svn=179" target="_blank">live demo</a> doesn't require any installation and can be used to connect to any SIP server using UDP, TCP or TLS transports.
</p>
<p>Short but not exhaustive list of supported features:</p>
<ul>
<li><b>Works on Chrome, Firefox, IE, Safari, Opera and Bowser</b></li>
<li>Audio / Video call</li>
<li><a href="https://code.google.com/p/sipml5/wiki/ScreenShare" target=_blank>Screen/Desktop sharing from Chrome to <b><u>any</u></b> SIP client</a></li>
<li>Instant messaging</li>
<li>Presence</li>
<li>Call Hold / Resume</li>
<li>Explicit Call transfer</li>
<li>Multi-line and multi-account</li>
<li>Dual-tone multi-frequency signaling (DTMF) using SIP INFO</li>
<li><a href="http://click2dial.org" target=_blank>Click-to-Call</a></li>
<li><a href="https://code.google.com/p/telepresence/" target=_blank>SIP TelePresence (Video Group chat)</a></li>
<li>3GPP IMS standards</li>
</ul>
<p><a class="btn" href="http://code.google.com/p/sipml5/" target="_blank">View details »</a></p>
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<tr><td align="center"><img alt="SIP Proxy architecture" src="http://webrtc2sip.googlecode.com/svn/trunk/documentation/images/architecture.png"/></td></tr>
<tr><td align="center"><i><u>sipML5 global solution architecture</u></i></td></tr>
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<tr><td><h2><a name="aMediaStack">Media Stack</a></h2></td></tr>
<tr><td>
<p>The media stack depends on <a href="http://en.wikipedia.org/wiki/WebRTC">WebRTC (Web Real Time Communication)</a> which is natively provided by the web browser.
<b>sipML5</b> should work on any web browser supporting WebRTC but we highly recommend using <a target=_blank href="https://www.google.com/intl/en/chrome/browser/">Google Chrome</a> or <a target=_blank href="http://nightly.mozilla.org/">Firefox Nightly</a> for testing. <br />
For <b>Safari</b>, <b>Firefox</b>, <b>Opera</b> and <b>IE</b> you will need to install <a href="http://code.google.com/p/webrtc4all/downloads/list" target="_blank"> webrtc4all</a> extension. <br />
</p>
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<tr><td><h2><a name="aInteroperability">Interoperability</a></h2></td></tr>
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<p>Using sipml5 and <a href="http://webrtc2sip.org">webrtc2sip.org</a> you can call any SIP-legacy endpoint or PSTN network. </p>
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<tr><td><h2><a name="aLicense">License</a></h2></td></tr>
<tr><td>
<p>The code is released under <a target=_blank href="http://opensource.org/licenses/BSD-3-Clause">BSD terms</a>. For more information: <a href="https://code.google.com/p/sipml5/wiki/License" target=_blank>https://code.google.com/p/sipml5/wiki/License</a> </p>
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<tr><td><h2><a name="aInteroperability">Programing with sipML5 API</a></h2></td></tr>
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<p>
The <a href="docgen/index.html">API</a> is designed with love to make it easy to develop rich and robust HTML5 applications in few lines of code. <br />
No need to know how SIP work to start writing your code. Using this <a href="docgen/index.html">API</a>, it will be a piece of cake to write HTML5 VoIP applications. <br />
Below, a very compact code showing how to initialize the engine, start the stack and make video call from <i>bob</i> to <i>alice</i> in <b>less than 15 lines</b>:
</p>
<pre><code>
<a href="docgen/symbols/SIPml.html#.init">SIPml.init</a>(
function(e){
var stack = new <a href="docgen/symbols/SIPml.Stack.html#constructor">SIPml.Stack</a>({realm: 'example.org', impi: 'bob', impu: 'sip:bob@example.org', password: 'mysecret',
events_listener: { events: 'started', listener: function(<a href="docgen/symbols/SIPml.Stack.Event.html">e</a>){
var callSession = stack.<a href="docgen/symbols/SIPml.Stack.html#newSession">newSession</a>('call-audiovideo', {
video_local: document.getElementById('video-local'), // <video id="video-local" .../>
video_remote: document.getElementById('video-remote'), // <video id="video-remote" .../>
audio_remote: document.getElementById('audio-remote') // <audio id="audio-remote" .../>
});
callSession.<a href="docgen/symbols/SIPml.Session.Call.html#call">call</a>('alice');
}
}
});
stack.<a href="docgen/symbols/SIPml.Stack.html#start">start</a>();
}
);
</code></pre>
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<td align="center"><i>Call between Google Chrome and iPad device</i></td>
<td align="center"><i>Call between Google Chrome and Android device</i></td>
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