Mesh-model is now handled on Signaling-Server.js

Server gives list of all participants as soon as you join a room. Added “autoRebootServerOnFailure=true” in the “config.json”. Which keeps server running all the time (auto reboots on failures)
muaz-khan · Oct 21, 2016 · 28f0f74 · 28f0f74
1 parent 2afabac
commit 28f0f74
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Showing 39 changed files with 296 additions and 271 deletions.
diff --git a/Signaling-Server.js b/Signaling-Server.js
@@ -278,6 +278,28 @@ module.exports = exports = function(app, socketCallback) {
             }
         }
 
+        function joinARoom(message) {
+            var roomInitiator = listOfUsers[message.remoteUserId];
+            var usersInARoom = roomInitiator.connectedWith;
+
+            var inviteTheseUsers = [roomInitiator.socket];
+            Object.keys(usersInARoom).forEach(function(key) {
+                inviteTheseUsers.push(usersInARoom[key]);
+            });
+
+            var keepUnique = [];
+            inviteTheseUsers.forEach(function(userSocket) {
+                if (userSocket.userid == socket.userid) return;
+                if (keepUnique.indexOf(userSocket.userid) != -1) {
+                    return;
+                }
+                keepUnique.push(userSocket.userid);
+
+                message.remoteUserId = userSocket.userid;
+                userSocket.emit(socketMessageEvent, message);
+            });
+        }
+
         var numberOfPasswordTries = 0;
         socket.on(socketMessageEvent, function(message, callback) {
             if (message.remoteUserId && message.remoteUserId === socket.userid) {
@@ -305,6 +327,11 @@ module.exports = exports = function(app, socketCallback) {
                             return;
                         }
                     }
+
+                    if (listOfUsers[message.remoteUserId]) {
+                        joinARoom(message);
+                        return;
+                    }
                 }
 
                 if (message.message.shiftedModerationControl) {
@@ -340,17 +367,21 @@ module.exports = exports = function(app, socketCallback) {
 
                 // if someone tries to join a person who is absent
                 if (message.message.newParticipationRequest) {
-                    var waitFor = 120; // 2 minutes
+                    var waitFor = 60 * 10; // 10 minutes
                     var invokedTimes = 0;
                     (function repeater() {
+                        if (typeof socket == 'undefined' || !listOfUsers[socket.userid]) {
+                            return;
+                        }
+
                         invokedTimes++;
                         if (invokedTimes > waitFor) {
                             socket.emit('user-not-found', message.remoteUserId);
                             return;
                         }
 
                         if (listOfUsers[message.remoteUserId] && listOfUsers[message.remoteUserId].socket) {
-                            onMessageCallback(message);
+                            joinARoom(message);
                             return;
                         }
 

diff --git a/config.json b/config.json
@@ -3,5 +3,6 @@
   "socketMessageEvent": "RTCMultiConnection-Message",
   "socketCustomEvent": "RTCMultiConnection-Custom-Message",
   "port": "9001",
-  "enableLogs": "true"
-}
+  "enableLogs": "true",
+  "autoRebootServerOnFailure": "true"
+}
diff --git a/demos/Audio+ScreenSharing.html b/demos/Audio+ScreenSharing.html
@@ -81,18 +81,15 @@ <h1>Audio+ScreenSharing using RTCMultiConnection</h1>
             (Install <a href="https://chrome.google.com/webstore/detail/screen-capturing/ajhifddimkapgcifgcodmmfdlknahffk" target="_blank">Chrome</a> / or / <a href="https://www.webrtc-experiment.com/store/firefox-extension/enable-screen-capturing.xpi" target="_blank">Firefox</a> extension) - Extensions are <a href="https://github.com/muaz-khan?tab=repositories">open-sourced</a>!
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
+        <script src="/dist/rmc3.min.js"></script>
+        <script src="/socket.io/socket.io.js"></script>
 
         <!-- capture screen from any HTTPs domain! -->
         <script src="https://cdn.webrtc-experiment.com:443/getScreenId.js"></script>
 
         <!-- custom layout for HTML5 audio/video elements -->
         <script src="https://cdn.webrtc-experiment.com/getMediaElement.js"></script>
 
-        <!-- socket.io for signaling -->
-        <script src="/socket.io/socket.io.js"></script>
-
         <script>
             // ......................................................
             // .......................UI Code........................

diff --git a/demos/Audio+Video+TextChat+FileSharing.html b/demos/Audio+Video+TextChat+FileSharing.html
@@ -84,16 +84,13 @@ <h1>Audio+Video+TextChat+FileSharing using RTCMultiConnection</h1>
             <div id="videos-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
+        <script src="/dist/rmc3.min.js"></script>
+        <script src="/socket.io/socket.io.js"></script>
 
         <!-- custom layout for HTML5 audio/video elements -->
         <script src="https://cdn.webrtc-experiment.com/getMediaElement.js"></script>
         <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
 
-        <!-- socket.io for signaling -->
-        <script src="/socket.io/socket.io.js"></script>
-
         <script>
             // ......................................................
             // .......................UI Code........................

diff --git a/demos/Audio-Conferencing.html b/demos/Audio-Conferencing.html
@@ -71,15 +71,11 @@ <h1>Audio Conferencing using RTCMultiConnection</h1>
             <div id="audios-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
+        <script src="/dist/rmc3.min.js"></script>
+        <script src="/socket.io/socket.io.js"></script>
 
-        <!-- custom layout for HTML5 audio/video elements -->
         <script src="https://cdn.webrtc-experiment.com/getMediaElement.js"></script>
 
-        <!-- socket.io for signaling -->
-        <script src="/socket.io/socket.io.js"></script>
-
         <script>
             // ......................................................
             // .......................UI Code........................

diff --git a/demos/Cross-Domain-Screen-Capturing.html b/demos/Cross-Domain-Screen-Capturing.html
@@ -93,17 +93,13 @@ <h1>Capture & Share Screen from any domain! using RTCMultiConnection</h1>
             <div id="videos-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
-
+        <script src="/dist/rmc3.min.js"></script>
         <script src="/dev/globals.js"></script>
+        <script src="/socket.io/socket.io.js"></script>
 
         <!-- capture screen from any HTTPs domain! -->
         <script src="https://cdn.webrtc-experiment.com:443/getScreenId.js"></script>
 
-        <!-- socket.io for signaling -->
-        <script src="/socket.io/socket.io.js"></script>
-
         <script>
             // ......................................................
             // .......................UI Code........................

diff --git a/demos/Disconnect+Rejoin.html b/demos/Disconnect+Rejoin.html
@@ -79,13 +79,11 @@ <h1>Disconnect+Rejoin using RTCMultiConnection</h1>
             <div id="videos-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
-        <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
-
-        <!-- socket.io for signaling -->
+        <script src="/dist/rmc3.min.js"></script>
         <script src="/socket.io/socket.io.js"></script>
 
+        <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
+
         <script>
             // ......................................................
             // .......................UI Code........................

diff --git a/demos/Files-Scalable-Broadcast.html b/demos/Files-Scalable-Broadcast.html
@@ -111,10 +111,9 @@ <h1><a href="https://github.com/muaz-khan/WebRTC-Scalable-Broadcast">WebRTC Scal
             Share camera with unlimited users using p2p methods! <a href="/demos/Video-Scalable-Broadcast.html">Video Scalable Broadcast</a> or <a href="/demos/Scalable-Broadcast.html">all-in-one scalable broadcast</a>.
         </blockquote>
 
+        <script src="/dist/rmc3.min.js"></script>
         <script src="/socket.io/socket.io.js"></script>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
         <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
 
         <script>

diff --git a/demos/Firebase-Demo.html b/demos/Firebase-Demo.html
@@ -75,13 +75,12 @@ <h1>Firebase Video Conferencing Demo using RTCMultiConnection</h1>
             <div id="videos-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
-        <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
+        <script src="/dist/rmc3.min.js"></script>
+        <script src="/dev/globals.js"></script>
+        <script src="/dev/FirebaseConnection.js"></script>
 
+        <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
         <script src="https://cdn.webrtc-experiment.com/firebase.js"></script>
-        <script src="../dev/globals.js"></script>
-        <script src="../dev/FirebaseConnection.js"></script>
 
         <script>
             // ......................................................

diff --git a/demos/Password-Protected-Rooms.html b/demos/Password-Protected-Rooms.html
@@ -80,13 +80,11 @@ <h1>Password+Protected+Rooms using RTCMultiConnection</h1>
             <div id="videos-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
-        <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
-
-        <!-- socket.io for signaling -->
+        <script src="/dist/rmc3.min.js"></script>
         <script src="/socket.io/socket.io.js"></script>
 
+        <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
+
         <script>
             // ......................................................
             // .......................UI Code........................

diff --git a/demos/Pre-recorded-Media-Streaming.html b/demos/Pre-recorded-Media-Streaming.html
@@ -89,16 +89,14 @@ <h1>Pre-recorded media streaming using RTCMultiConnection</h1>
             <div id="videos-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
+        <script src="/dist/rmc3.min.js"></script>
+        <script src="/socket.io/socket.io.js"></script>
+
         <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
 
         <!-- custom layout for HTML5 audio/video elements -->
         <script src="https://cdn.webrtc-experiment.com/getMediaElement.js"></script>
 
-        <!-- socket.io for signaling -->
-        <script src="/socket.io/socket.io.js"></script>
-
         <script>
             // ......................................................
             // .......................UI Code........................

diff --git a/demos/PubNub-Demo.html b/demos/PubNub-Demo.html
@@ -73,13 +73,13 @@ <h1>PubNub Video Conferencing Demo using RTCMultiConnection</h1>
             <div id="videos-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
+        <script src="/dist/rmc3.min.js"></script>
+        <script src="/dev/globals.js"></script>
+        <script src="/dev/PubNubConnection.js"></script>
+
         <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
 
         <script src="https://cdn.webrtc-experiment.com/view/websocket.js"></script>
-        <script src="../dev/globals.js"></script>
-        <script src="../dev/PubNubConnection.js"></script>
 
         <script>
             // ......................................................

diff --git a/demos/Scalable-Broadcast.html b/demos/Scalable-Broadcast.html
@@ -119,16 +119,9 @@ <h1><a href="https://github.com/muaz-khan/WebRTC-Scalable-Broadcast">WebRTC Scal
             Share files with unlimited users using p2p methods! <a href="/demos/Files-Scalable-Broadcast.html">Files Scalable Broadcast</a>
         </blockquote>
 
+        <script src="/dist/rmc3.min.js"></script>
         <script src="/socket.io/socket.io.js"></script>
-
-        <script>
-            // skipping xirsys servers
-            window.getExternalIceServers = false;
-        </script>
-
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com/rmc3.min.js"></script>
-
+
         <!-- <script src="https://cdn.webrtc-experiment.com/RecordRTC.js"></script> -->
         <script>
             // recording is disabled because it is resulting for browser-crash

diff --git a/demos/Scalable-Screen-Broadcast.html b/demos/Scalable-Screen-Broadcast.html
@@ -118,22 +118,16 @@ <h1><a href="https://github.com/muaz-khan/WebRTC-Scalable-Broadcast">WebRTC Scal
             Requirements: <a href="https://chrome.google.com/webstore/detail/screen-capturing/ajhifddimkapgcifgcodmmfdlknahffk">install this Chrome extension</a> or <a href="https://addons.mozilla.org/en-US/firefox/addon/enable-screen-capturing/">this Firefox extension</a>.
         </blockquote>
 
+        <script src="/dist/rmc3.min.js"></script>
         <script src="/socket.io/socket.io.js"></script>
 
-        <script>
-            // skipping xirsys servers
-            window.getExternalIceServers = false;
-        </script>
-
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
-
         <!-- capture screen from any HTTPs domain! -->
         <script src="https://cdn.webrtc-experiment.com:443/getScreenId.js"></script>
 
-        <script src="https://cdn.webrtc-experiment.com/RecordRTC.js"></script>
+        <!-- <script src="https://cdn.webrtc-experiment.com/RecordRTC.js"></script> -->
         <script>
             // recording is disabled because it is resulting for browser-crash
+            // if you enable below line, please also uncomment above "RecordRTC.js"
             var enableRecordings = false;
 
             var connection = new RTCMultiConnection(null, {

diff --git a/demos/Scalable-Screen-plus-Audio-Broadcast.html b/demos/Scalable-Screen-plus-Audio-Broadcast.html
@@ -124,22 +124,16 @@ <h1><a href="https://github.com/muaz-khan/WebRTC-Scalable-Broadcast">WebRTC Scal
             Requirements: <a href="https://chrome.google.com/webstore/detail/screen-capturing/ajhifddimkapgcifgcodmmfdlknahffk">install this Chrome extension</a> or <a href="https://addons.mozilla.org/en-US/firefox/addon/enable-screen-capturing/">this Firefox extension</a>.
         </blockquote>
 
+        <script src="/dist/rmc3.min.js"></script>
         <script src="/socket.io/socket.io.js"></script>
 
-        <script>
-            // skipping xirsys servers
-            window.getExternalIceServers = false;
-        </script>
-
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
-
         <!-- capture screen from any HTTPs domain! -->
         <script src="https://cdn.webrtc-experiment.com:443/getScreenId.js"></script>
 
-        <script src="https://cdn.webrtc-experiment.com/RecordRTC.js"></script>
+        <!-- <script src="https://cdn.webrtc-experiment.com/RecordRTC.js"></script> -->
         <script>
             // recording is disabled because it is resulting for browser-crash
+            // if you enable below line, please also uncomment above "RecordRTC.js"
             var enableRecordings = false;
 
             var connection = new RTCMultiConnection(null, {

diff --git a/demos/StreamHasData.html b/demos/StreamHasData.html
@@ -87,12 +87,8 @@ <h1>Check if StreamHasData using RTCMultiConnection</h1>
             <div id="videos-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
-
+        <script src="/dist/rmc3.min.js"></script>
         <script src="/dev/StreamHasData.js"></script>
-
-        <!-- socket.io for signaling -->
         <script src="/socket.io/socket.io.js"></script>
 
         <script>

diff --git a/demos/TextChat+FileSharing.html b/demos/TextChat+FileSharing.html
@@ -74,13 +74,11 @@ <h1>TextChat+FileSharing using RTCMultiConnection</h1>
             </div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
-        <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
-
-        <!-- socket.io for signaling -->
+        <script src="/dist/rmc3.min.js"></script>
         <script src="/socket.io/socket.io.js"></script>
 
+        <script src="https://cdn.webrtc-experiment.com:443/FileBufferReader.js"></script>
+
         <script>
             // ......................................................
             // .......................UI Code........................

diff --git a/demos/Video-Conferencing.html b/demos/Video-Conferencing.html
@@ -88,8 +88,7 @@ <h1>Video Conferencing using RTCMultiConnection</h1>
             <div id="videos-container"></div>
         </section>
 
-        <!-- <script src="/dist/rmc3.min.js"></script> -->
-        <script src="https://cdn.webrtc-experiment.com:443/rmc3.min.js"></script>
+        <script src="/dist/rmc3.min.js"></script>
 
         <!-- custom layout for HTML5 audio/video elements -->
         <script src="https://cdn.webrtc-experiment.com/getMediaElement.js"></script>