From 95b76ae4922589c8302a3ea7f454c19645038b34 Mon Sep 17 00:00:00 2001
From: Joseph Frazier <joseph@onsip.com>
Date: Sat, 14 Jun 2014 11:44:31 -0400
Subject: [PATCH] Use SIP.js master branch and specify MediaHandler at runtime

---
 .gitmodules                    |   2 +-
 sip.js                         |   2 +-
 www/index.html                 |  11 +-
 www/js/PhoneRTCMediaHandler.js | 283 +++++++++++++++++++++++++++++++++
 4 files changed, 295 insertions(+), 3 deletions(-)
 create mode 100644 www/js/PhoneRTCMediaHandler.js

diff --git a/.gitmodules b/.gitmodules
index eeca764..9f0c632 100644
--- a/.gitmodules
+++ b/.gitmodules
@@ -1,3 +1,3 @@
 [submodule "sip.js"]
 	path = sip.js
-	url = https://github.com/joseph-onsip/SIP.js.git
+	url = https://github.com/onsip/SIP.js.git
diff --git a/sip.js b/sip.js
index 314e745..c21846c 160000
--- a/sip.js
+++ b/sip.js
@@ -1 +1 @@
-Subproject commit 314e745c28839f9aa7649f2c8533fce3a1b83cf7
+Subproject commit c21846c14645dd4214e8d4f3c7c91b6649d9932c
diff --git a/www/index.html b/www/index.html
index cda539a..cb1da7c 100644
--- a/www/index.html
+++ b/www/index.html
@@ -42,18 +42,27 @@ <h1>Apache Cordova</h1>
         <script type="text/javascript" src="cordova.js"></script>
         <script type="text/javascript" src="js/index.js"></script>
         <script type="text/javascript" src="js/sip.js"></script>
+        <script type="text/javascript" src="js/PhoneRTCMediaHandler.js"></script>
         <script type="text/javascript">
             app.initialize();
 
             var $ = document.querySelector.bind(document);
 
+            // SIP.js refuses to work otherwise
+            SIP.WebRTC.isSupported = true;
+
             $('#phone').addEventListener('submit', function (e) {
               if (window.session) {
                 console.warn('only one call at a time');
                 return;
               }
 
-              window.ua = window.ua || new SIP.UA({traceSip: true});
+              window.ua = window.ua || new SIP.UA({
+                traceSip: true,
+                mediaHandlerFactory: function defaultFactory (session, options) {
+                  return new PhoneRTCMediaHandler(session, options);
+                }
+              });
               window.session = window.ua.invite($('#target').value);
 
               // adapted from http://stackoverflow.com/a/5150556
diff --git a/www/js/PhoneRTCMediaHandler.js b/www/js/PhoneRTCMediaHandler.js
new file mode 100644
index 0000000..6b35147
--- /dev/null
+++ b/www/js/PhoneRTCMediaHandler.js
@@ -0,0 +1,283 @@
+var PhoneRTCMediaHandler = function(session, options) {
+  var events = [
+  ];
+  options = options || {};
+
+  this.logger = session.ua.getLogger('sip.invitecontext.mediahandler', session.id);
+  this.session = session;
+  this.localMedia = null;
+  this.ready = true;
+  this.mediaStreamManager = SIP.WebRTC.MediaStreamManager.cast(options.mediaStreamManager);
+  this.audioMuted = false;
+  this.videoMuted = false;
+
+  // old init() from here on
+  var idx, length, server,
+    servers = [],
+    stunServers = options.stunServers || null,
+    turnServers = options.turnServers || null,
+    config = this.session.ua.configuration;
+  this.RTCConstraints = options.RTCConstraints || {};
+
+  if (!stunServers) {
+    stunServers = config.stunServers;
+  }
+
+  if(!turnServers) {
+    turnServers = config.turnServers;
+  }
+
+  /* Change 'url' to 'urls' whenever this issue is solved:
+   * https://code.google.com/p/webrtc/issues/detail?id=2096
+   */
+  servers.push({'url': stunServers});
+
+  length = turnServers.length;
+  for (idx = 0; idx < length; idx++) {
+    server = turnServers[idx];
+    servers.push({
+      'url': server.urls,
+      'username': server.username,
+      'credential': server.password
+    });
+  }
+
+  this.initEvents(events);
+
+  this.phonertc = {};
+};
+
+PhoneRTCMediaHandler.prototype = Object.create(SIP.MediaHandler.prototype, {
+// Functions the session can use
+  isReady: {writable: true, value: function isReady () {
+    return this.ready;
+  }},
+
+  close: {writable: true, value: function close () {
+    this.logger.log('closing PeerConnection');
+    // have to check signalingState since this.close() gets called multiple times
+    // TODO figure out why that happens
+    if(this.peerConnection && this.peerConnection.signalingState !== 'closed') {
+      this.peerConnection.close();
+
+      if(this.localMedia) {
+        this.mediaStreamManager.release(this.localMedia);
+      }
+    }
+  }},
+
+  /**
+   * @param {Function} onSuccess
+   * @param {Function} onFailure
+   * @param {SIP.WebRTC.MediaStream | (getUserMedia constraints)} [mediaHint]
+   *        the MediaStream (or the constraints describing it) to be used for the session
+   */
+  getDescription: {writable: true, value: function getDescription (onSuccess, onFailure, mediaHint) {
+                    onFailure = onFailure;
+                    mediaHint = mediaHint;
+    if (!this.phonertc.role) {
+      this.phonertcCall('caller');
+    }
+
+    var pcDelay = 2000;
+    setTimeout(function () {
+      var sdp = this.phonertc.localSdp;
+      if (this.phonertc.role !== 'caller') {
+        sdp = sdp.replace('a=setup:actpass', 'a=setup:passive');
+      }
+      sdp = sdp.replace(/a=crypto.*\r\n/g, '');
+      onSuccess(sdp);
+    }.bind(this), pcDelay);
+  }},
+
+  phonertcSendMessageCallback: {writable: true, value: function phonertcSendMessageCallback (data) {
+    this.logger.log("XXX phonertcSendMessageCallback: " + JSON.stringify(data, null, 2));
+    if (['offer', 'answer'].indexOf(data.type) > -1) {
+      this.phonertc.localSdp = data.sdp;
+    }
+    else if (data.type === 'candidate') {
+      this.phonertc.localSdp += data.candidate;
+    }
+  }},
+
+  phonertcCall: {writable: true, value: function phonertcCall (role) {
+    this.logger.log("XXX phonertcCall: " + role);
+    this.phonertc.role = role;
+    cordova.plugins.phonertc.call({
+      isInitator: role === 'caller', // Caller or callee?
+      turn: {
+        host: 'turn:turn.example.com:3478',
+        username: 'user',
+        password: 'pass'
+      },
+      sendMessageCallback: this.phonertcSendMessageCallback.bind(this),
+      answerCallback: function () {
+        window.alert('Callee answered!');
+      },
+      disconnectCallback: function () {
+        window.alert('Call disconnected!');
+      }
+    });
+  }},
+
+  /**
+  * Message reception.
+  * @param {String} type
+  * @param {String} sdp
+  * @param {Function} onSuccess
+  * @param {Function} onFailure
+  */
+  setDescription: {writable: true, value: function setDescription (sdp, onSuccess, onFailure) {
+    function setRemoteDescription (type, sdp) {
+      this.logger.log("XXX setRemoteDescription: " + type + "\n" + sdp);
+      cordova.plugins.phonertc.receiveMessage({type: type, sdp: sdp});
+      onSuccess();
+    }
+
+    if (!this.phonertc.role) {
+      this.phonertcCall('callee');
+      var pcDelay = 2000;
+      setTimeout(setRemoteDescription.bind(this, 'offer', sdp), pcDelay);
+    }
+    else if (this.phonertc.role = 'caller') {
+      setRemoteDescription.call(this, 'answer', sdp);
+    }
+    else {
+      this.logger.error('XXX setDescription called, but this.phonertc.role = ' + this.phonertc.role);
+      onFailure();
+    }
+  }},
+
+// Functions the session can use, but only because it's convenient for the application
+  isMuted: {writable: true, value: function isMuted () {
+    return {
+      audio: this.audioMuted,
+      video: this.videoMuted
+    };
+  }},
+
+  mute: {writable: true, value: function mute (options) {
+          options = options;
+  }},
+
+  unmute: {writable: true, value: function unmute (options) {
+            options = options;
+  }},
+
+  hold: {writable: true, value: function hold () {
+    this.toggleMuteAudio(true);
+    this.toggleMuteVideo(true);
+  }},
+
+  unhold: {writable: true, value: function unhold () {
+    if (!this.audioMuted) {
+      this.toggleMuteAudio(false);
+    }
+
+    if (!this.videoMuted) {
+      this.toggleMuteVideo(false);
+    }
+  }},
+
+// Functions the application can use, but not the session
+  getLocalStreams: {writable: true, value: function getLocalStreams () {
+    var pc = this.peerConnection;
+    if (pc && pc.signalingState === 'closed') {
+      this.logger.warn('peerConnection is closed, getLocalStreams returning []');
+      return [];
+    }
+    return (pc.getLocalStreams && pc.getLocalStreams()) ||
+      pc.localStreams || [];
+  }},
+
+  getRemoteStreams: {writable: true, value: function getRemoteStreams () {
+    var pc = this.peerConnection;
+    if (pc && pc.signalingState === 'closed') {
+      this.logger.warn('peerConnection is closed, getRemoteStreams returning []');
+      return [];
+    }
+    return(pc.getRemoteStreams && pc.getRemoteStreams()) ||
+      pc.remoteStreams || [];
+  }},
+
+// Internal functions
+  hasOffer: {writable: true, value: function hasOffer (where) {
+    var offerState = 'have-' + where + '-offer';
+    return this.peerConnection.signalingState === offerState;
+    // TODO consider signalingStates with 'pranswer'?
+  }},
+
+  createOfferOrAnswer: {writable: true, value: function createOfferOrAnswer (onSuccess, onFailure, constraints) {
+    var self = this;
+
+    function readySuccess () {
+      var sdp = self.peerConnection.localDescription.sdp;
+
+      sdp = SIP.Hacks.Chrome.needsExplicitlyInactiveSDP(sdp);
+
+      self.ready = true;
+      onSuccess(sdp);
+    }
+
+    function onSetLocalDescriptionSuccess() {
+      if (self.peerConnection.iceGatheringState === 'complete' && self.peerConnection.iceConnectionState === 'connected') {
+        readySuccess();
+      } else {
+        self.onIceCompleted = function() {
+          self.logger.log('ICE Gathering Completed');
+          self.onIceCompleted = undefined;
+          readySuccess();
+        };
+      }
+    }
+
+    function methodFailed (methodName, e) {
+      self.logger.error('peerConnection.' + methodName + ' failed');
+      self.logger.error(e);
+      self.ready = true;
+      onFailure(e);
+    }
+
+    self.ready = false;
+
+    var methodName = self.hasOffer('remote') ? 'createAnswer' : 'createOffer';
+
+    self.peerConnection[methodName](
+      function(sessionDescription){
+        self.peerConnection.setLocalDescription(
+          sessionDescription,
+          onSetLocalDescriptionSuccess,
+          methodFailed.bind(null, 'setLocalDescription')
+        );
+      },
+      methodFailed.bind(null, methodName),
+      constraints
+    );
+  }},
+
+  addStream: {writable: true, value: function addStream (stream, onSuccess, onFailure, constraints) {
+    try {
+      this.peerConnection.addStream(stream, constraints);
+    } catch(e) {
+      this.logger.error('error adding stream');
+      this.logger.error(e);
+      onFailure(e);
+      return;
+    }
+
+    onSuccess();
+  }},
+
+  toggleMuteHelper: {writable: true, value: function toggleMuteHelper (trackGetter, mute) {
+                      trackGetter = trackGetter;
+                      mute = mute;
+  }},
+
+  toggleMuteAudio: {writable: true, value: function toggleMuteAudio (mute) {
+    this.toggleMuteHelper('getAudioTracks', mute);
+  }},
+
+  toggleMuteVideo: {writable: true, value: function toggleMuteVideo (mute) {
+    this.toggleMuteHelper('getVideoTracks', mute);
+  }}
+});