RTCMultiConnection-v1.3 is ready for tests!

You can test streams renegotiation features; mute/unmute; bandwidth;
direct messages; and much more.

To read what’s new in v1.3 and what you can do:

https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection

1. You can send direct messages
2. You can renegotiate streams among all users
3. You can renegotiate streams directly between two unique users
4. You can mute/unmute any stream
5. You can set bandwidth for audio/video streams

The only issue in the moment is “screen-sharing in one-way” is not
working. This issue occurs only with renegotiated streams.

When someone Mute or UnMute stream; currently unable to alert him about
the action; so he can show a “poster” in the place of “video” box. It
will be fixed in the next commits.

An advance demo will be coming soon (in a few days); till that you can
try this “simple” demo:

https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3/

Author: muaz-khan

Diff for: README.md

@@ -10,19 +10,19 @@ These demos/experiments are entirely client-side; i.e. no server installation ne
 
 ### Libraries
 
-1. [RTCMultiConnection.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection) — A JavaScript library for streams renegotiation and sharing; multi-session establishment and much more.
-2. [DataChannel.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel) — A JavaScript library for data/file/text sharing!
-3. [RecordRTC.js](https://webrtc-experiment.appspot.com/RecordRTC/) — A library to record audio/video streams / [Demo](https://webrtc-experiment.appspot.com/RecordRTC/)
-4. [RTCall.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCall) — A library for Browser-to-Browser audio-only calling / [Demo](https://webrtc-experiment.appspot.com/RTCall/)
+1. [RTCMultiConnection.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection) — [Demos](https://webrtc-experiment.appspot.com/#RTCMultiConnection)
+2. [DataChannel.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel) — [Demos](https://webrtc-experiment.appspot.com/#DataChannel)
+3. [RecordRTC.js](https://webrtc-experiment.appspot.com/RecordRTC/) — [Demo](https://webrtc-experiment.appspot.com/RecordRTC/)
+4. [RTCall.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCall) — [Demo](https://webrtc-experiment.appspot.com/RTCall/)
 
 ----
 
 #### WebRTC Signaling Gateways
 
 Following signaling gateways can work with each and every [WebRTC Experiment](https://webrtc-experiment.appspot.com/)!
 
-1. [WebSocket over Node.js](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/websocket-over-nodejs) — [Demo](http://websocket-over-nodejs.jit.su/)
-2. [Socket.io over Node.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socketio-over-nodejs) — [Demo](http://webrtc-signaling.jit.su/)
+1. [Socket.io over Node.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socketio-over-nodejs) — [Demo](http://webrtc-signaling.jit.su/)
+2. [WebSocket over Node.js](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/websocket-over-nodejs) — [Demo](http://websocket-over-nodejs.jit.su/)
 
 ----
 

Diff for: RTCMultiConnection/README.md

@@ -1,7 +1,3 @@
-`RTCMultiConnection-v1.3` is still under developement; see a [testing demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3/). You're recommended to try [v1.2](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.2-and-earlier.md) until `v1.3` gets ready.
-
-----
-
 #### [RTCMultiConnection](https://webrtc-experiment.appspot.com/#RTCMultiConnection): A simple library to handle advance tasks!
 
 Supports features like
@@ -153,16 +149,6 @@ connection.streams['stream-id'].unmute({
 
 ----
 
-#### `removeStream`
-
-```javascript
-connection.streams['stream-id'].removeStream();
-```
-
-Above line will remove a stream and renegotiate the peer connection.
-
-----
-
 #### Detect users presence
 
 ```javascript

Diff for: RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html

@@ -1,5 +1,6 @@
 <!DOCTYPE html>
-<html lang="en">    
+<html lang="en">
+    
     <head>
         <title>RTCMultiConnection-v1.3.js testing demo! ® Muaz Khan</title>
         <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
@@ -8,18 +9,18 @@
         <link rel="author" type="text/html" href="https://plus.google.com/100325991024054712503">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-		<script>
+        <script>
             var hash = window.location.hash.replace('#', '');
             if (!hash.length) {
                 location.href = location.href + '#' + ((Math.random() * new Date().getTime()).toString(36).toUpperCase().replace(/\./g, '-'));
                 location.reload();
             }
         </script>
-		<script src="/firebase.js"></script>
+        <script src="/firebase.js"></script>
         <script src="/RTCMultiConnection-v1.3.js"></script>
     </head>
 
-    <body>        
+    <body>
         <button id="new-session">Open New Session</button>
         <input type="checkbox" id="audio-renegotiation" checked>
         <label for="audio-renegotiation">Renegotiate Audio</label>
@@ -28,13 +29,11 @@
         <input type="checkbox" id="screen-renegotiation">
         <label for="screen-renegotiation">Renegotiate Screen</label>
         <input type="checkbox" id="one-way">
-        <label for="one-way">Is One Way?</label>
-        <a href="https://webrtc-experiment.appspot.com/">Goto "WebRTC Experiments & Demos" Homepage</a>
+        <label for="one-way">Is One Way?</label> <a href="https://webrtc-experiment.appspot.com/">Goto "WebRTC Experiments & Demos" Homepage</a>
 
         <div id="local-video-container" class="local-video-container"></div>
         <div id="remote-videos-container" class="remote-videos-container"></div>
-        
-        <h2>How it works?</h2>
+         <h2>How it works?</h2>
 
         <p>Renegotiation means using pre-created peer connections to add additional streams by renegotiating offer/answer session descriptions.</p>
         <p>In this demo; at first time; only data connection is opened. Later on, same peer connections are used to renegotiate audio, video or screen streams.</p>
@@ -93,6 +92,29 @@ <h2>How it works?</h2>
                 div = document.createElement('div');
                 div.className = 'video-container';
                 div.id = e.userid || 'You';
+
+                button = document.createElement('button');
+                button.innerHTML = 'Mute';
+                button.id = e.streamid;
+                button.onclick = function () {
+                    button = this;
+                    button.disabled = true;
+                    streamid = button.id;
+
+                    if (button.innerHTML == 'Mute') {
+                        connection.streams[streamid].mute();
+                        button.innerHTML = 'UnMute';
+                    } else {
+                        connection.streams[streamid].unmute();
+                        button.innerHTML = 'Mute';
+                    }
+
+                    setTimeout(function () {
+                        button.disabled = false;
+                    }, 1000);
+                };
+                div.appendChild(button);
+
                 div.appendChild(e.mediaElement);
 
                 h2 = document.createElement('h2');
@@ -112,7 +134,9 @@ <h2>How it works?</h2>
                 -webkit-user-select:none;
                 font-family:'Segoe UI Light';
             }
-            html{overflow: hidden;}
+            html {
+                overflow: hidden;
+            }
             .local-video-container {
                 border: 2px solid red;
                 width:40%;
@@ -229,4 +253,5 @@ <h2>How it works?</h2>
             }
         </script>
     </body>
+
 </html>