v2.0.5

js/modules/webrtc/qbRTCPeerConnection.js

@@ -45,6 +45,7 @@ RTCPeerConnection.prototype.init = function(delegate, userID, sessionID, type) {
   /** We use this timer interval to dial a user - produce the call requests each N seconds. */
   this.dialingTimer = null;
   this.answerTimeInterval = 0;
+  this.statsReportTimer = null;
 
   /** timer to detect network blips */
   this.reconnectTimer = 0;
@@ -54,6 +55,7 @@ RTCPeerConnection.prototype.init = function(delegate, userID, sessionID, type) {
 
 RTCPeerConnection.prototype.release = function(){
   this._clearDialingTimer();
+  this._clearStatsReportTimer();
 
   if(this.signalingState !== 'closed'){
     this.close();
@@ -174,16 +176,19 @@ RTCPeerConnection.prototype.onIceCandidateCallback = function(event) {
 
 /** handler of remote media stream */
 RTCPeerConnection.prototype.onAddRemoteStreamCallback = function(event) {
+  var self = this;
+
   if (typeof this.delegate._onRemoteStreamListener === 'function'){
     this.delegate._onRemoteStreamListener(this.userID, event.stream);
   }
+
+  self._getStatsWrap();
 };
 
 RTCPeerConnection.prototype.onIceConnectionStateCallback = function() {
   var newIceConnectionState = this.iceConnectionState;
-
-  Helpers.trace("onIceConnectionStateCallback: " + this.iceConnectionState);
 
+  Helpers.trace("onIceConnectionStateCallback: " + this.iceConnectionState);
 
   /**
    * read more about all states:
@@ -226,6 +231,45 @@ RTCPeerConnection.prototype.onIceConnectionStateCallback = function() {
 /**
  * PRIVATE
  */
+ RTCPeerConnection.prototype._clearStatsReportTimer = function(){
+   if(this.statsReportTimer){
+     clearInterval(this.statsReportTimer);
+     this.statsReportTimer = null;
+   }
+ };
+
+RTCPeerConnection.prototype._getStatsWrap = function() {
+  var self = this,
+      selector = self.delegate.callType == 1 ? self.getLocalStreams()[0].getVideoTracks()[0] : self.getLocalStreams()[0].getAudioTracks()[0],
+      statsReportInterval;
+
+  if (!config.webrtc && !config.webrtc.statsReportTimeInterval) {
+    return;
+  }
+
+  if (isNaN(+config.webrtc.statsReportTimeInterval)) {
+     Helpers.traceError('statsReportTimeInterval (' + config.webrtc.statsReportTimeInterval + ') must be integer.');
+     return;
+  }
+
+  statsReportInterval = config.webrtc.statsReportTimeInterval * 1000;
+
+  var _statsReportCallback = function() {
+    _getStats(self, selector,
+      function (results) {
+        self.delegate._onCallStatsReport(self.userID, results);
+      },
+      function errorLog(err) {
+        Helpers.traceError("_getStats error. " + err.name + ": " + err.message);
+      }
+    );
+   };
+
+   Helpers.trace('Stats tracker has been started.');
+
+   self.statsReportTimer = setInterval(_statsReportCallback, statsReportInterval);
+ };
+
 RTCPeerConnection.prototype._clearWaitingReconnectTimer = function() {
   if(this.waitingReconnectTimeoutCallback){
     Helpers.trace('_clearWaitingReconnectTimer');
@@ -292,4 +336,40 @@ RTCPeerConnection.prototype._startDialingTimer = function(extension, withOnNotAn
   _dialingCallback(extension, withOnNotAnswerCallback, true);
 };
 
-module.exports = RTCPeerConnection;
+/**
+ * PRIVATE
+ */
+ function _getStats(peer, selector, successCallback, errorCallback) {
+  /**
+   * http://stackoverflow.com/questions/25069419/webrtc-getstat-api-set-up
+   */
+   if (navigator.mozGetUserMedia) {
+     peer.getStats(selector,
+       function (res) {
+         var items = [];
+         res.forEach(function (result) {
+             items.push(result);
+         });
+         successCallback(items);
+       },
+       errorCallback
+     );
+   } else {
+     peer.getStats(function (res) {
+       var items = [];
+       res.result().forEach(function (result) {
+         var item = {};
+         result.names().forEach(function (name) {
+             item[name] = result.stat(name);
+         });
+         item.id = result.id;
+         item.type = result.type;
+         item.timestamp = result.timestamp;
+         items.push(item);
+       });
+       successCallback(items);
+     });
+   }
+ }
+
+module.exports = RTCPeerConnection;

js/modules/webrtc/qbWebRTCClient.js

@@ -80,6 +80,7 @@ WebRTCClient.prototype._createAndStoreSession = function(sessionID, callerID, op
   newSession.onRemoteStreamListener = this.onRemoteStreamListener;
   newSession.onSessionConnectionStateChangedListener = this.onSessionConnectionStateChangedListener;
   newSession.onSessionCloseListener = this.onSessionCloseListener;
+  newSession.onCallStatsReport = this.onCallStatsReport;
 
   this.sessions[newSession.ID] = newSession;
   return newSession;

js/modules/webrtc/qbWebRTCSession.js

@@ -9,6 +9,7 @@
  * - onRemoteStreamListener(session, userID, stream)
  * - onSessionConnectionStateChangedListener(session, userID, connectionState)
  * - onSessionCloseListener(session)
+ * - onCallStatsReport(session, userId, stats)
  */
 
 var config = require('../../qbConfig');
@@ -360,6 +361,24 @@ WebRTCSession.prototype.stop = function(extension) {
   self._close();
 };
 
+/**
+ * [function close connection with user]
+ * @param  {[type]} userId [id of user]
+ */
+WebRTCSession.prototype.closeConnection = function(userId) {
+  var self = this,
+    peer = this.peerConnections[userId];
+
+  if(peer) {
+    peer.release();
+
+    self._closeSessionIfAllConnectionsClosed();
+  } else {
+    Helpers.traceWarn('Not found connection with user (' + userId + ')');
+  }
+};
+
+
 /**
  * Update a call
  * @param {array} A map with custom parameters
@@ -594,6 +613,12 @@ WebRTCSession.prototype._onRemoteStreamListener = function(userID, stream) {
   }
 };
 
+WebRTCSession.prototype._onCallStatsReport = function(userId, stats) {
+  if (typeof this.onCallStatsReport === 'function'){
+    Utils.safeCallbackCall(this.onCallStatsReport, this, userId, stats);
+  }
+};
+
 WebRTCSession.prototype._onSessionConnectionStateChangedListener = function(userID, connectionState) {
   var self = this;
 

js/qbConfig.js

@@ -3,6 +3,10 @@
  *
  * Configuration Module
  *
+ * NOTE:
+ *  - config.webrtc.statsReportTimeInterval [integer, sec]:
+ *  could add listener onCallStatsReport(session, userId, bytesReceived) if
+ *  want to get stats (bytesReceived) about peer every X sec;
  */
 
 var config = {
@@ -26,6 +30,7 @@ var config = {
     answerTimeInterval: 60,
     dialingTimeInterval: 5,
     disconnectTimeInterval: 30,
+    statsReportTimeInterval: 10,
     iceServers: [
       {
         'url': 'stun:stun.l.google.com:19302'

package.json

@@ -44,7 +44,7 @@
     "crypto-js": "3.1.2-2",
     "jquery": "^2.2.0",
     "request": "^2.48.0",
-    "strophe": "^1.2.3",
+    "strophe": "~1.2.4",
     "xml2js": "^0.4.13"
   },
   "devDependencies": {

samples/chat/index.html

@@ -147,7 +147,7 @@ <h5 class="col-md-12 col-sm-12 col-xs-12" id="all_occupants"></h5>
   <script src="https://cdnjs.cloudflare.com/ajax/libs/jquery-timeago/1.4.1/jquery.timeago.min.js" type="text/javascript"></script>
   <script src="https://cdnjs.cloudflare.com/ajax/libs/twitter-bootstrap/3.3.5/js/bootstrap.min.js" type="text/javascript"></script>
 
-  <script src="../../quickblox.min.js"></script>
+  <script src="../../quickblox.js"></script>
   <script src="js/config.js"></script>
   <script src="js/connection.js"></script>
   <script src="js/messages.js"></script>

samples/webrtc/config.js

@@ -5,7 +5,7 @@
    *
    * Possible options:
    * https://examples.com?users=prod
-   * https://examples.com?users=dev 
+   * https://examples.com?users=dev
    * https://examples.com - for qa by default
    */
   var usersQuery = _getQueryVar('users');
@@ -15,7 +15,8 @@
     webrtc: {
       answerTimeInterval: 30,
       dialingTimeInterval: 5,
-      disconnectTimeInterval: 30
+      disconnectTimeInterval: 30,
+      statsReportTimeInterval: 5
     }
   };
 
@@ -231,4 +232,4 @@
   window.CONFIG = CONFIG;
   window.QBUsers = QBUsers;
   window.MESSAGES = MESSAGES;
-}(window));
+}(window));