Release SIPml API 1.0.0 and public documentation

git-svn-id: http://sipml5.googlecode.com/svn/trunk@144 63d8a0da-676c-2731-e3aa-b417aa27da68
samsonlarsson · Dec 17, 2012 · a3acfee · a3acfee
1 parent a09dafd
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diff --git a/SIPml-api.js b/SIPml-api.js
@@ -0,0 +1,24 @@
+/*
+* Copyright (C) 2012 Doubango Telecom <http://www.doubango.org>
+* License: GPLv3
+* This file is part of Open Source sipML5 solution <http://www.sipml5.org>
+*/
+
+// IMPORTANT: /!\ DO NOT EDIT THIS FILE /!\
+
+// ================== DEBUG CODE =============== //
+var __b_debug_mode = true;
+var tag_hdr = document.getElementsByTagName('head')[0];
+['SIPml.js', 'src/tinySIP/src/tsip_api.js'].forEach(function (file) {
+    var tag_script = document.createElement('script');
+    tag_script.setAttribute('type', 'text/javascript');
+    tag_script.setAttribute('src', file + "?svn=20");
+    tag_hdr.appendChild(tag_script);
+});
+if (window.console) {
+    window.console.warn("YOUR ARE USING DEBUG CODE. PLEASE USE CODE UNDER 'release' FOLDER.");
+}
+
+// ================== RELEASE CODE =============== //
+// http://code.google.com/p/sipml5/wiki/FAQ#How_to_reduce_the_size_of_the_scripts_before_deploying
+//
diff --git a/SIPml.js b/SIPml.js
diff --git a/call.htm b/call.htm
diff --git a/docgen.index.html b/docgen.index.html
diff --git a/docgen.sh b/docgen.sh
@@ -0,0 +1,19 @@
+java \
+-Djsdoc.dir=./jsdoc-toolkit \
+-jar ./jsdoc-toolkit/jsrun.jar \
+./jsdoc-toolkit/app/run.js \
+-t=./jsdoc-toolkit/templates/jsdoc \
+-r=4 \
+./SIPml.js \
+-d=docgen \
+-D="title:SIPml Library" \
+-D"index:files"
+
+cp docgen.index.html docgen/index.html
+
+sed -i 's/="assets/="..\/assets/g' docgen/index.html
+sed -i 's/="docgen\//="/g' docgen/index.html
+sed -i 's/="images/="..\/images/g' docgen/index.html
+
+sed -i 's/Namespace /AnonymousClass /g' docgen/symbols/SIPml.Session.Configuration.html
+sed -i 's/Namespace /AnonymousClass /g' docgen/symbols/SIPml.Stack.Configuration.html
diff --git a/google-closure-compiler.jar b/google-closure-compiler.jar
diff --git a/images/classes.bmp b/images/classes.bmp
diff --git a/images/classes.dia b/images/classes.dia
diff --git a/images/rectangle_256x256.png b/images/rectangle_256x256.png
diff --git a/index.html b/index.html
@@ -20,6 +20,9 @@
         padding-top: 80px;
         padding-bottom: 40px;
       }
+      tr.spaceUnder > td {
+        padding-bottom: 2em;
+      }
     </style>
     <link href="./assets/css/bootstrap-responsive.css" rel="stylesheet" />
 
@@ -50,9 +53,10 @@
           <a class="brand" href="./index.html">Doubango Telecom</a>
           <div class="nav-collapse">
             <ul class="nav">
-              <li class="active"><a href="index.html?svn=13">Home</a></li>
+              <li class="active"><a href="index.html?svn=20">Home</a></li>
               <li><a href="http://code.google.com/p/sipml5/source/checkout" target="_blank">Source code</a></li>
-              <li><a href="contact.htm?svn=13">Contact</a></li>
+              <li><a href="docgen/index.html?svn=20">Programmer's Guide</a></li>
+              <li><a href="contact.htm">Contact</a></li>
             </ul>
           </div><!--/.nav-collapse -->
         </div>
@@ -67,71 +71,106 @@ <h2>World's first HTML5 SIP client</h2>
         <p>This is the world's first open source HTML5 SIP client entirely written in javascript for integration in social networks (FaceBook, Twitter, Google+), online games, e-commerce websites... 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=13" target="_blank">Enjoy our live demo &raquo;</a></p>
+        <p><a class="btn btn-primary btn-large" href="call.htm?svn=20" target="_blank">Enjoy our live demo &raquo;</a></p>
       </div>
 
-      <!-- Example row of columns -->
-      <div class="row-fluid">
-        <div class="span4">
-          <h3>Javascript SIP/SDP stack</h3>
-           <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-02">draft-ibc-sipcore-sip-websocket-02</a>.
-           This <a href="call.htm?svn=13" 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>Audio / Video call</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>
-          </ul>
-          <p><a class="btn" href="http://code.google.com/p/sipml5/" target="_blank">View details &raquo;</a></p>
-        </div>
-        <div class="span4">
-          <h3>Media Stack</h3>
-           <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 href="https://www.google.com/intl/en/chrome/browser/">Google Chrome 21.0.1180.60 m</a> or later 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 />
-           For more information on how to enable WebRTC on chrome, please visit <a href="http://code.google.com/p/sipml5/wiki/Enable_WebRTC_On_Chrome" target="_blank"> this page</a>.
-           </p>
-          <p><a class="btn" href="http://code.google.com/p/sipml5/wiki/Enable_WebRTC_On_Chrome" target="_blank">View details &raquo;</a></p>
-       </div>
-         <div class="span4">
-          <h3>Interoperability</h3>
-          <p>The implementation follows the standards defined by the <a href="http://tools.ietf.org/wg/rtcweb/" target="_blank">RTCWeb</a> and <a href="http://www.w3.org/TR/webrtc/" target="_blank">W3C</a> working groups. You can call any SIP client (mobile devices, browser, desktop client, set-top box...).</p>
-          <p>For interoperability with rich SIP clients we recommend:</p>
-          <ul>
-            <li><a href="http://code.google.com/p/boghe/">Boghe</a>: Windows XP, Vista and 7</li>
-            <li><a href="http://code.google.com/p/imsdroid/">IMSDroid</a>: Android</li>
-            <li><a href="http://code.google.com/p/idoubs/">iDoubs</a>: iOS (iPad, iPhone and iPod Touch) and MAC OS X</li>
-          </ul>
-          All these clients are developed using <a href="http://www.doubango.org" target="_blank"> Doubango IMS Framework</a>.
-          <p><a class="btn" href="http://code.google.com/p/sipml5/wiki/Calling_SIP_clients" target="_blank">View details &raquo;</a></p>
-        </div>
-       </div>
-
-       <br />
-
-       <table>
-           <tr>
-                <td>
-                    <iframe id="ytplayer" type="text/html" width="512" height="312"
-                      src="http://www.youtube.com/embed/ro3FFNx7d-g?autoplay=0&origin=http://sipml5.org"
-                      frameborder="0"> </iframe>&nbsp;&nbsp;
-                </td>
-                <td>
-                    <iframe id="Iframe1" type="text/html" width="512" height="312"
-                      src="http://www.youtube.com/embed/7HEMyxAnATI?autoplay=0&origin=http://sipml5.org"
-                      frameborder="0"> </iframe>
-                </td>
-           </tr>
-           <tr>
-                <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>
-           </tr>
-       </table>
+
+      <table>
+          <tr class="spaceUnder"><td>
+                <table>
+                  <tr><td><h2><a name="aSipSdpStack">Javascript SIP/SDP stack</a></h2></td></tr>
+                  <tr class="spaceUnder"><td>
+                       <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=13" 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>Audio / Video call</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>
+                      </ul>
+                      <p><a class="btn" href="http://code.google.com/p/sipml5/" target="_blank">View details &raquo;</a></p>       
+                   </td></tr>
+                  <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>
+              </table>
+         </td></tr>
+
+         <tr><td>
+                <table>
+                  <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 href="https://www.google.com/intl/en/chrome/browser/">Google Chrome 21.0.1180.60 m</a> or later 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>     
+                   </td></tr>
+              </table>
+         </td></tr>
+
+         <tr><td>
+                <table>
+                  <tr><td><h2><a name="aInteroperability">Interoperability</a></h2></td></tr>
+                  <tr><td>
+                     <p>Using sipml5 and <a href="http://webrtc2sip.org">webrtc2sip.org</a> you can call any SIP-legacy endpoint or PSTN network. </p>
+                  </td></tr>
+              </table>
+         </td></tr>
+
+         <tr class="spaceUnder"><td>
+                <table>
+                  <tr><td><h2><a name="aInteroperability">Programing with sipML5 API</a></h2></td></tr>
+                  <tr class="spaceUnder"><td>
+                     <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:[email protected]', 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_remote: document.getElementById('video-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>
+                  </td></tr>
+              </table>
+         </td></tr>
+
+     </table>
+
+    <table>
+        <tr>
+            <td>
+                <iframe id="ytplayer" type="text/html" width="512" height="312"
+                    src="http://www.youtube.com/embed/ro3FFNx7d-g?autoplay=0&origin=http://sipml5.org"
+                    frameborder="0"> </iframe>&nbsp;&nbsp;
+            </td>
+            <td>
+                <iframe id="Iframe1" type="text/html" width="512" height="312"
+                    src="http://www.youtube.com/embed/7HEMyxAnATI?autoplay=0&origin=http://sipml5.org"
+                    frameborder="0"> </iframe>
+            </td>
+        </tr>
+        <tr>
+            <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>
+        </tr>
+    </table>
 
     <hr />
     <g:plus href="https://plus.google.com/107177632824809995368" rel="author"></g:plus>