Migrating Apprtc to use new TURN service which supports time-limited TURN credentials.

Review URL: https://webrtc-codereview.appspot.com/1291004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3773 4adac7df-926f-26a2-2b94-8c16560cd09d

samples/js/apprtc/apprtc.py

@@ -315,6 +315,7 @@ class MainPage(webapp2.RequestHandler):
     min_re = self.request.get('minre')
     max_re = self.request.get('maxre')
     hd_video = self.request.get('hd')
+    turn_url = 'https://computeengineondemand.appspot.com/'
     if hd_video.lower() == 'true':
       min_re = '1280x720'
     ts_pwd = self.request.get('tp')
@@ -374,6 +375,7 @@ class MainPage(webapp2.RequestHandler):
 
     room_link = base_url + '?r=' + room_key
     room_link = append_url_arguments(self.request, room_link)
+    turn_url = turn_url + 'turn?' + 'username=' + user + '&key=4080218913'
     token = create_channel(room, user, token_timeout)
     pc_config = make_pc_config(stun_server, turn_server, ts_pwd)
     pc_constraints = make_pc_constraints(compat)
@@ -387,7 +389,8 @@ class MainPage(webapp2.RequestHandler):
                        'pc_config': json.dumps(pc_config),
                        'pc_constraints': json.dumps(pc_constraints),
                        'offer_constraints': json.dumps(offer_constraints),
-                       'media_constraints': json.dumps(media_constraints)
+                       'media_constraints': json.dumps(media_constraints),
+                       'turn_url': turn_url
                       }
     if unittest:
       target_page = 'test/test_' + unittest + '.html'

samples/js/apprtc/index.html

@@ -112,10 +112,13 @@
   var remoteStream;
   var channel;
   var channelReady = false;
+  var turnReady = false;
   var pc;
   var socket;
   var initiator = {{ initiator }};
   var started = false;
+  var pc_config = {{ pc_config|safe }};
+  var pc_constraints = {{ pc_constraints|safe }};
   // Set up audio and video regardless of what devices are present.
   var sdpConstraints = {'mandatory': {
                           'OfferToReceiveAudio':true, 
@@ -133,6 +136,7 @@
     // NOTE: AppRTCClient.java searches & parses this line; update there when
     // changing here.
     openChannel('{{ token }}');
+    requestTurn('{{ turn_url }}');
     doGetUserMedia();
   }
 
@@ -148,6 +152,35 @@
     socket = channel.open(handler);
   }
 
+  function requestTurn(turn_url) {
+    var turnExists = false;
+    for (var i in pc_config.iceServers) {
+      if (pc_config.iceServers[i].url.substr(0, 5) == 'turn:') {
+        turnExists = true;
+        turnReady = true;
+        break;
+      }
+    }
+    if (!turnExists) {
+      // No turn server. Get one from computeengineondemand.appspot.com:
+      xmlhttp = new XMLHttpRequest();
+      xmlhttp.onreadystatechange = onTurnResult;
+      xmlhttp.open("GET", turn_url, true);
+      xmlhttp.send();
+    }
+  }
+
+  function onTurnResult() {
+    if (xmlhttp.readyState == 4 && xmlhttp.status == 200) {
+      var turnServer = JSON.parse(xmlhttp.responseText);
+      pc_config.iceServers.push({
+        "url": "turn:" + turnServer.username + "@" + turnServer.turn,
+        "credential": turnServer.password
+      });
+      turnReady = true;
+    }
+  }
+
   function resetStatus() {
     if (!initiator) {
       setStatus("Waiting for someone to join: <a href=\"{{ room_link }}\">{{ room_link }}</a>");
@@ -170,13 +203,11 @@
     }
   }
 
-  function createPeerConnection() {
-    var pc_config = {{ pc_config|safe }};
-    var pc_constraints = {{ pc_constraints|safe }};
+  function createPeerConnection() {    
     // Force the use of a number IP STUN server for Firefox.
     if (webrtcDetectedBrowser == "firefox") {
       pc_config = {"iceServers":[{"url":"stun:23.21.150.121"}]};
-    }    
+    }
     try {
       // Create an RTCPeerConnection via the polyfill (adapter.js).
       pc = new RTCPeerConnection(pc_config, pc_constraints);
@@ -189,13 +220,12 @@
       alert("Cannot create RTCPeerConnection object; WebRTC is not supported by this browser.");
         return;
     }
-
     pc.onaddstream = onRemoteStreamAdded;
     pc.onremovestream = onRemoteStreamRemoved;
   }
 
   function maybeStart() {
-    if (!started && localStream && channelReady) {
+    if (!started && localStream && channelReady && turnReady) {
       setStatus("Connecting...");
       console.log("Creating PeerConnection.");
       createPeerConnection();