From 25af0379537778a855e9328cd1b3075f66a9e227 Mon Sep 17 00:00:00 2001
From: Muaz Khan <muazkh@gmail.com>
Date: Sat, 20 Jul 2013 06:44:33 +0500
Subject: [PATCH] =?UTF-8?q?WebRTC=20Experiments=20=E2=80=94=20https://www.?=
 =?UTF-8?q?webrtc-experiment.com/=20=E2=80=94=20updated.?=
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

---
 AudioVideoRecorder/README.md                  |  14 +-
 AudioVideoRecorder/index.html                 |  12 +-
 DataChannel/DataChannel.js                    |   9 +-
 DataChannel/README.md                         |  12 +-
 DataChannel/auto-session-establishment.html   |  14 +-
 DataChannel/index.html                        |  14 +-
 LICENSE                                       |   2 +-
 Pluginfree-Screen-Sharing/README.md           |  19 +-
 Pluginfree-Screen-Sharing/conference-ui.js    |  37 +-
 Pluginfree-Screen-Sharing/index.html          |  18 +-
 Pre-recorded-Media-Streaming/README.md        |  30 +-
 Pre-recorded-Media-Streaming/index.html       |  14 +-
 README.md                                     | 196 ++++----
 RTCDataConnection/README.md                   |   9 +-
 .../RTCDataConnection-Manual-Test.html        |   4 +-
 .../RTCDataConnection-Simple-Test.html        |   4 +-
 RTCDataConnection/RTCDataConnection-Test.html |   4 +-
 RTCMultiConnection/README.md                  |  54 +-
 ...RTCMultiConnection-v1.3-and-socket.io.html |   4 +-
 .../RTCMultiConnection-v1.3-demo.html         |   6 +-
 .../RTCMultiConnection-Demos/all-in-one.html  |  14 +-
 .../audio-conferencing-data-sharing.html      |  14 +-
 .../audioconferencing.html                    |  14 +-
 .../RTCMultiConnection-Demos/bandwidth.html   |  14 +-
 .../group-file-sharing-plus-text-chat.html    |  14 +-
 .../join-with-or-without-camera.html          |  14 +-
 ...a-transmission-plus-videoconferencing.html |  12 +-
 ...lishment-plus-extra-data-transmission.html |  14 +-
 .../multi-session-establishment.html          |  14 +-
 .../one-to-one-filesharing.html               |  14 +-
 .../screen-sharing.html                       |  14 +-
 .../users-ejection.html                       |  14 +-
 .../video-broadcasting.html                   |  14 +-
 .../videoconferencing.html                    |  10 +-
 .../RTCMultiConnection-v1.2-and-earlier.md    |  39 +-
 RTCMultiConnection/RTCMultiConnection-v1.3.md |  43 +-
 .../All-in-One.html                           |  14 +-
 .../Renegotiation.html                        |   6 +-
 .../Video-Conferencing.html                   |  10 +-
 .../multi-streams-attachment.html             |   6 +-
 .../All-in-One.html                           |  14 +-
 .../Renegotiation.html                        |   6 +-
 .../Video-Conferencing.html                   |  10 +-
 .../RTCMultiConnection-v1.5-experimental.md   |  16 +-
 RTCMultiConnection/index.html                 | 383 +++++++-------
 RTCPeerConnection/README.md                   | 307 +++++++++---
 RTCPeerConnection/RTCPeerConnection-v1.5.js   | 258 +++++-----
 RTCall/README.md                              |  35 +-
 RTCall/index.html                             |  16 +-
 RecordRTC/README.md                           |  14 +-
 RecordRTC/index.html                          |  20 +-
 audio-broadcast/README.md                     |  12 +-
 audio-broadcast/index.html                    |  16 +-
 broadcast/README.md                           |  14 +-
 broadcast/index.html                          |  16 +-
 chat-hangout/README.md                        |  14 +-
 chat-hangout/hangout-ui.js                    |  67 +--
 chat-hangout/hangout.js                       | 195 +++-----
 chat-hangout/index.html                       |  16 +-
 chat/README.md                                |  12 +-
 demos/README.md                               |  22 +-
 demos/client-side-datachannel.html            |  12 +-
 demos/client-side.html                        |   8 +-
 demos/remote-stream-recording.html            | 171 +++++++
 demos/screen-and-video-from-single-peer.html  |   8 +-
 .../How-to-Broadcast-Screen-using-WebRTC.html |   6 +-
 docs/README.md                                |  27 +-
 docs/how-file-broadcast-works.html            |  14 +-
 docs/how-to-WebRTC-video-conferencing.html    |   8 +-
 ...-to-broadcast-video-using-RTCWeb-APIs.html |   8 +-
 docs/how-to-install-tabCapture-extension.html |  10 +-
 docs/how-to-share-audio-only-streams.html     |   8 +-
 docs/how-to-use-plugin-free-calls.html        |   6 +-
 docs/how-to-use-rtcdatachannel.html           |   8 +-
 .../how-to-use-rtcpeerconnection-js-v1.1.html |  10 +-
 docs/rtc-datachannel-for-beginners.html       |  14 +-
 docs/webrtc-for-beginners.html                |   8 +-
 docs/webrtc-for-newbies.html                  |   8 +-
 experimental/README.md                        |   4 +-
 .../mozCaptureStreamUntilEnded/README.md      |   6 +-
 .../mozCaptureStreamUntilEnded/index.html     |   8 +-
 .../remote-media-stream-attachment/README.md  |   6 +-
 .../remote-media-stream-attachment/index.html |   8 +-
 experimental/remote-stream-recording.html     | 122 +++--
 file-hangout/README.md                        |  16 +-
 file-hangout/hangout-ui.js                    |  83 ++--
 file-hangout/hangout.js                       | 371 +++++++-------
 file-hangout/index.html                       |  18 +-
 file-sharing/README.md                        |  30 +-
 file-sharing/index.html                       |  12 +-
 meeting/README.md                             |  26 +-
 meeting/index.html                            |  12 +-
 one-to-many-audio-broadcasting/README.md      |  29 +-
 one-to-many-audio-broadcasting/index.html     |  12 +-
 one-to-many-video-broadcasting/README.md      |  28 +-
 one-to-many-video-broadcasting/index.html     |  12 +-
 p2p-share/README.md                           |  17 +-
 part-of-screen-sharing/README.md              |  18 +-
 part-of-screen-sharing/index.html             | 468 +++++++++---------
 .../realtime-chat/README.md                   |  15 +-
 .../realtime-chat/how-this-work.html          |   2 +-
 .../realtime-chat/index.html                  |   2 +-
 .../README.md                                 |  18 +-
 .../index.html                                |  31 +-
 realtime-pluginfree-calls/README.md           |   6 +-
 realtime-pluginfree-calls/index.html          |  41 +-
 screen-broadcast/README.md                    |  30 +-
 screen-broadcast/index.html                   |  18 +-
 screen-sharing/README.md                      |  36 +-
 screen-sharing/index.html                     |  14 +-
 socket.io/README.md                           |  12 +-
 socket.io/index.html                          |  16 +-
 socket.io/rtclib.js                           | 204 ++++----
 socket.io/ui.js                               |  83 ++--
 socketio-over-nodejs/README.md                |  30 +-
 socketio-over-nodejs/Signaling-Concepts.md    |  22 +-
 .../webrtc-signaling/README.md                |   6 +-
 .../static/RTCMultiConnection-v1.4/index.html |   2 +-
 .../static/RTCMultiConnection/index.html      |   4 +-
 .../webrtc-signaling/static/text-chat.html    |   2 +-
 .../RTCPeerConnection-v1.5.js                 | 258 +++++-----
 .../video-conferencing/conference-ui.js       |  77 +--
 .../static/video-conferencing/conference.js   | 169 ++++---
 .../static/video-conferencing/index.html      |   8 +-
 text-chat/README.md                           |  28 +-
 text-chat/index.html                          |  12 +-
 video-conferencing/README.md                  |  18 +-
 video-conferencing/conference-ui.js           |  76 +--
 video-conferencing/conference.js              | 189 +++----
 video-conferencing/index.html                 |  16 +-
 webrtc-broadcasting/README.md                 |  12 +-
 webrtc-broadcasting/broadcast-ui.js           |  39 +-
 webrtc-broadcasting/index.html                |  16 +-
 websocket-over-nodejs/README.md               |  22 +-
 .../RTCPeerConnection-v1.5.js                 | 259 +++++-----
 .../video-conferencing/index.html             |   8 +-
 websocket/README.md                           |  12 +-
 websocket/index.html                          |  16 +-
 138 files changed, 3148 insertions(+), 2568 deletions(-)
 create mode 100644 demos/remote-stream-recording.html

diff --git a/AudioVideoRecorder/README.md b/AudioVideoRecorder/README.md
index 607f7c70..877bd6f7 100644
--- a/AudioVideoRecorder/README.md
+++ b/AudioVideoRecorder/README.md
@@ -1,4 +1,4 @@
-##### Audio+Video Recording using MediaRecorder / [Demo](https://webrtc-experiment.appspot.com/AudioVideoRecorder/)
+##### Audio+Video Recording using MediaRecorder / [Demo](https://www.webrtc-experiment.com/AudioVideoRecorder/)
 
 Only audio-relevant parts are supported in the moment. Audio+Video recording coming soon.
 
@@ -6,10 +6,16 @@ Support? Current/Latest Firefox Nightly (ONLY). Understood? Download from: http:
 
 =
 
+##### Note
+
+There is another relevant project: https://github.com/streamproc/MediaStreamRecorder
+
+=
+
 ##### How to use AudioVideoRecorder?
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/AudioVideoRecorder.js"></script>
+<script src="https://www.webrtc-experiment.com/AudioVideoRecorder.js"></script>
 ```
 
 =
@@ -47,7 +53,7 @@ AudioVideoRecorder({
 
 ##### Browser Support
 
-[AudioVideoRecorder.js](https://webrtc-experiment.appspot.com/AudioVideoRecorder/) works fine on following web-browsers:
+[AudioVideoRecorder.js](https://www.webrtc-experiment.com/AudioVideoRecorder/) works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -57,4 +63,4 @@ AudioVideoRecorder({
 
 ##### License
 
-[AudioVideoRecorder.js](https://webrtc-experiment.appspot.com/AudioVideoRecorder/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[AudioVideoRecorder.js](https://www.webrtc-experiment.com/AudioVideoRecorder/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/AudioVideoRecorder/index.html b/AudioVideoRecorder/index.html
index d1a52030..3ee2071c 100644
--- a/AudioVideoRecorder/index.html
+++ b/AudioVideoRecorder/index.html
@@ -91,7 +91,7 @@
 
             ::-moz-selection { background: #ccc; }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/AudioVideoRecorder.js"> </script>
+        <script src="https://www.webrtc-experiment.com/AudioVideoRecorder.js"> </script>
     </head>
     <body>
         <h1>
@@ -103,7 +103,7 @@ <h1>
             Copyright © 2013 <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a>&lt;<a
                                                                                                          href="http://twitter.com/muazkh" target="_blank">@muazkh</a>&gt;.</p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <div id="audios-container">
         </div>
@@ -112,7 +112,7 @@ <h1>
             navigator.mozGetUserMedia(mediaConstraints, onMediaSuccess, onMediaError);
 
             function onMediaSuccess(stream) {
-                // https://webrtc-experiment.appspot.com/AudioVideoRecorder.js
+                // https://www.webrtc-experiment.com/AudioVideoRecorder.js
                 AudioVideoRecorder({
                     stream: stream,
                     mimeType: 'audio/ogg',
@@ -147,7 +147,7 @@ <h3>
             <strong style="color: red;">Limitation</strong>: Only audio-recording is available
             (<span style="color: rgb(187, 192, 213);">...audio+video recording coming soon</span>)</h3>
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/AudioVideoRecorder.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/AudioVideoRecorder.js"&gt;&lt;/script&gt;
 
 AudioVideoRecorder({
 
@@ -185,12 +185,12 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span> 2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span> 2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                     href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
diff --git a/DataChannel/DataChannel.js b/DataChannel/DataChannel.js
index 34eda78e..cba854ab 100644
--- a/DataChannel/DataChannel.js
+++ b/DataChannel/DataChannel.js
@@ -235,7 +235,7 @@
         var self = { };
         var that = this;
 
-        self.userToken = root.userid || uniqueToken();
+        self.userToken = root.userid = root.userid || uniqueToken();
         self.sockets = [];
         self.socketObjects = { };
 
@@ -819,9 +819,10 @@
                 var message = content[uuid].join('');
                 if (data.isobject) message = JSON.parse(message);
 
-                // latency detection
+                // bug: latency detection must be fixed
+                // https://github.com/muaz-khan/WebRTC-Experiment/issues/63#issuecomment-21083575
                 var receivingTime = new Date().getTime();
-                var latency = receivingTime - data.sendingTime;
+                var latency = Math.abs(receivingTime - data.sendingTime);
 
                 if (onmessage) onmessage(message, userid, latency);
 
@@ -861,7 +862,7 @@
             credential: 'homeo'
         };
 
-        iceServers = {
+        var iceServers = {
             iceServers: options.iceServers || [STUN]
         };
 
diff --git a/DataChannel/README.md b/DataChannel/README.md
index 93bf739a..61b2267e 100644
--- a/DataChannel/README.md
+++ b/DataChannel/README.md
@@ -1,4 +1,4 @@
-##### [DataChannel.js](https://webrtc-experiment.appspot.com/DataChannel.js) : A JavaScript wrapper library for RTCDataChannel APIs / [Demos](https://webrtc-experiment.appspot.com/#DataChannel)
+##### [DataChannel.js](https://www.webrtc-experiment.com/DataChannel.js) : A JavaScript wrapper library for RTCDataChannel APIs / [Demos](https://www.webrtc-experiment.com/#DataChannel)
 
 DataChannel.js is a JavaScript library useful to write many-to-many i.e. group file/data sharing or text chat applications. Its syntax is easier to use and understand. It highly simplifies complex tasks like any or all user rejection/ejection; direct messages delivery; and more.
 
@@ -22,7 +22,7 @@ DataChannel.js is a JavaScript library useful to write many-to-many i.e. group f
 ##### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/DataChannel.js"></script>
+<script src="https://www.webrtc-experiment.com/DataChannel.js"></script>
 ```
 
 =
@@ -289,10 +289,10 @@ channel.connect('channel-name');
 
 =
 
-##### [Demos using DataChannel.js](https://webrtc-experiment.appspot.com/#DataChannel)
+##### [Demos using DataChannel.js](https://www.webrtc-experiment.com/#DataChannel)
 
-1. [DataChannel basic demo](https://webrtc-experiment.appspot.com/DataChannel/)
-2. [Auto Session Establishment and Users presence detection](https://webrtc-experiment.appspot.com/DataChannel/auto-session-establishment/)
+1. [DataChannel basic demo](https://www.webrtc-experiment.com/DataChannel/)
+2. [Auto Session Establishment and Users presence detection](https://www.webrtc-experiment.com/DataChannel/auto-session-establishment/)
 
 =
 
@@ -310,4 +310,4 @@ channel.connect('channel-name');
 
 ##### License
 
-[DataChannel.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[DataChannel.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/DataChannel/auto-session-establishment.html b/DataChannel/auto-session-establishment.html
index b55766f6..f98492a2 100644
--- a/DataChannel/auto-session-establishment.html
+++ b/DataChannel/auto-session-establishment.html
@@ -206,13 +206,13 @@
         document.createElement('footer');
     </script>
 
-    <script src="https://webrtc-experiment.appspot.com/DataChannel.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
+    <script src="https://www.webrtc-experiment.com/DataChannel.js"></script>
+    <script src="https://www.webrtc-experiment.com/firebase.js"></script>
 </head>
 
 <body>
 <article>
-<a href="https://webrtc-experiment.appspot.com/"
+<a href="https://www.webrtc-experiment.com/"
    style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑
     WEBRTC EXPERIMENTS</a>
 
@@ -227,7 +227,7 @@ <h1>Auto Session Establishment using <a href="https://github.com/muaz-khan/WebRT
         href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
 </p>
 <section class="plusone-gplus">
-    <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+    <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
 </section>
 <table style="border-left: 1px solid black; width: 100%;">
     <tr>
@@ -341,7 +341,7 @@ <h2 style="display: block; font-size: 1em; text-align: center;">Share Files</h2>
 <h2>Getting started with WebRTC <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel"
                                    target="_blank">DataChannel</a></h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/DataChannel.js" target="_blank">https://webrtc-experiment.appspot.com/DataChannel.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/DataChannel.js" target="_blank">https://www.webrtc-experiment.com/DataChannel.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var channel = new <strong>DataChannel</strong>('default-channel');
 
@@ -490,7 +490,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -498,6 +498,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/DataChannel/index.html b/DataChannel/index.html
index d1da413e..df7d06c9 100644
--- a/DataChannel/index.html
+++ b/DataChannel/index.html
@@ -183,13 +183,13 @@
         document.createElement('footer');
     </script>
 
-    <script src="https://webrtc-experiment.appspot.com/DataChannel.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
+    <script src="https://www.webrtc-experiment.com/DataChannel.js"></script>
+    <script src="https://www.webrtc-experiment.com/firebase.js"></script>
 </head>
 
 <body>
 <article>
-<a href="https://webrtc-experiment.appspot.com/"
+<a href="https://www.webrtc-experiment.com/"
    style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
     WEBRTC EXPERIMENTS</a>
 
@@ -205,7 +205,7 @@ <h1>File Sharing + Text Chat using WebRTC <a
         href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
 </p>
 <section class="plusone-gplus">
-    <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+    <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
 </section>
 <section>
     <h2>Open Data Channel</h2>
@@ -325,7 +325,7 @@ <h2 style="display: block; font-size: 1em; text-align: center;">Share Files</h2>
 <h2>Getting started with WebRTC <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel"
                                    target="_blank">DataChannel</a></h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/DataChannel.js" target="_blank">https://webrtc-experiment.appspot.com/DataChannel.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/DataChannel.js" target="_blank">https://www.webrtc-experiment.com/DataChannel.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var channel = new <strong>DataChannel</strong>();
 
@@ -439,7 +439,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -447,6 +447,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/LICENSE b/LICENSE
index 9a2330de..6c4b6b12 100644
--- a/LICENSE
+++ b/LICENSE
@@ -9,4 +9,4 @@ The above copyright notice and this permission notice shall be included in all c
 
 THE SOFTWARE IS PROVIDED �AS IS�, WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 
-MIT: https://webrtc-experiment.appspot.com/licence/
+MIT: https://www.webrtc-experiment.com/licence/
diff --git a/Pluginfree-Screen-Sharing/README.md b/Pluginfree-Screen-Sharing/README.md
index 86f33ce2..c52e986c 100644
--- a/Pluginfree-Screen-Sharing/README.md
+++ b/Pluginfree-Screen-Sharing/README.md
@@ -1,7 +1,9 @@
-#### WebRTC plugin free screen sharing / [Demo](https://webrtc-experiment.appspot.com/Pluginfree-Screen-Sharing/)
+#### WebRTC plugin free screen sharing / [Demo](https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/)
 
 Share entire screen directly without any single installation!
 
+=
+
 #### Enable screen capture support in getUserMedia()
 
 1. Open `chrome://flags` in the latest chrome (canary/beta). 
@@ -21,24 +23,29 @@ navigator.webkitGetUserMedia({
 }, onstreaming, onfailure);
 ```
 
-There is another experiment: [WebRTC Tab Sharing using experimental tabCapture APIs](https://webrtc-experiment.appspot.com/screen-broadcast/)
+There is another experiment: [WebRTC Tab Sharing using experimental tabCapture APIs](https://www.webrtc-experiment.com/screen-broadcast/)
+
+=
 
 #### What about Desktop Sharing?
 
 It is a big wish to share desktop using RTCWeb peer connection APIs but unfortunately currently it is not possible.
 
 Current experiment is using chrome screen sharing APIs which is allows end-users just **view the screen**....nothing else!
+=
 
 #### Browser Support
 
-[WebRTC plugin free screen sharing](https://webrtc-experiment.appspot.com/Pluginfree-Screen-Sharing/) experiment works fine on following web-browsers:
+[WebRTC plugin free screen sharing](https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
-| ------------- |:-------------|
+| ------------- |-------------|
 | Firefox | [Stable](http://www.mozilla.org/en-US/firefox/new/) / [Aurora](http://www.mozilla.org/en-US/firefox/aurora/) / [Nightly](http://nightly.mozilla.org/) |
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
-| Internet Explorer / IE | [Chrome Frame](http://www.google.com/chromeframe) |
+| Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
+
+=
 
 #### License
 
-[WebRTC plugin free screen sharing](https://webrtc-experiment.appspot.com/Pluginfree-Screen-Sharing/) experiment is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC plugin free screen sharing](https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/) experiment is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/Pluginfree-Screen-Sharing/conference-ui.js b/Pluginfree-Screen-Sharing/conference-ui.js
index 6db09378..f8ae4f19 100644
--- a/Pluginfree-Screen-Sharing/conference-ui.js
+++ b/Pluginfree-Screen-Sharing/conference-ui.js
@@ -1,17 +1,34 @@
-var config = {
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing
+
+var config = {
     openSocket: function (config) {
-        var channel = config.channel || location.hash.replace('#', '') || 'pluginfree-screen-sharing';
-        var socket = new Firebase('https://rtcweb.firebaseIO.com/' + channel);
+        var SIGNALING_SERVER = 'https://www.webrtc-experiment.com:8553/',
+            defaultChannel = location.hash.substr(1) || 'Pluginfree-Screen-Sharing';
+
+        var channel = config.channel || defaultChannel;
+        var sender = Math.round(Math.random() * 999999999) + 999999999;
+
+        io.connect(SIGNALING_SERVER).emit('new-channel', {
+            channel: channel,
+            sender: sender
+        });
+
+        var socket = io.connect(SIGNALING_SERVER + channel);
         socket.channel = channel;
-        socket.on('child_added', function (data) {
-            config.onmessage && config.onmessage(data.val());
+        socket.on('connect', function() {
+            if (config.callback) config.callback(socket);
         });
-        socket.send = function (data) {
-            this.push(data);
+
+        socket.send = function(message) {
+            socket.emit('message', {
+                sender: sender,
+                data: message
+            });
         };
-        config.onopen && setTimeout(config.onopen, 1);
-        socket.onDisconnect().remove();
-        return socket;
+
+        socket.on('message', config.onmessage);
     },
     onRemoteStream: function (media) {
         var video = media.video;
diff --git a/Pluginfree-Screen-Sharing/index.html b/Pluginfree-Screen-Sharing/index.html
index 398b93ec..eb74af34 100644
--- a/Pluginfree-Screen-Sharing/index.html
+++ b/Pluginfree-Screen-Sharing/index.html
@@ -190,7 +190,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
         <h1>WebRTC Plugin-free Screen Sharing</h1>
         <p>
@@ -199,7 +199,7 @@ <h1>WebRTC Plugin-free Screen Sharing</h1>
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <div id="participants"></div>
@@ -223,10 +223,10 @@ <h2>
             </tr>
         </table>
         <table id="rooms-list" class="visible"></table>
-        <script src='https://webrtc-experiment.appspot.com/firebase.js'></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/Pluginfree-Screen-Sharing/conference.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/Pluginfree-Screen-Sharing/conference-ui.js"></script>
+        <script src='https://www.webrtc-experiment.com/socket.io.js'></script>
+        <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
+        <script src="https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/conference.js"></script>
+        <script src="https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/conference-ui.js"></script>
 
         <br />
         <br />
@@ -268,7 +268,7 @@ <h2>To use code in your own site, you must understand following limitations:</h2
             Remember, recursive cascade images or blurred screen is chrome's implementation issues. It will be solved soon.
             <br />
             <br />
-            <code>mandatory: {chromeMediaSource: '<span style="color: red;">tab</span>'}</code> can only be useful in chrome extensions. See <a href="https://webrtc-experiment.appspot.com/screen-broadcast/" target="_blank">Tab sharing using tabCapture APIs</a>.
+            <code>mandatory: {chromeMediaSource: '<span style="color: red;">tab</span>'}</code> can only be useful in chrome extensions. See <a href="https://www.webrtc-experiment.com/screen-broadcast/" target="_blank">Tab sharing using tabCapture APIs</a>.
         </blockquote>
 
         <br />
@@ -291,13 +291,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/Pre-recorded-Media-Streaming/README.md b/Pre-recorded-Media-Streaming/README.md
index 88fb65a7..a4f89040 100644
--- a/Pre-recorded-Media-Streaming/README.md
+++ b/Pre-recorded-Media-Streaming/README.md
@@ -1,20 +1,24 @@
-#### Pre-recorded media streaming / [Demo](https://webrtc-experiment.appspot.com/Pre-recorded-Media-Streaming/)
+#### Pre-recorded media streaming / [Demo](https://www.webrtc-experiment.com/Pre-recorded-Media-Streaming/)
 
 1. Streaming pre-recorded video (media file)
 2. Currently, using `Firebase` for streaming chunks of data because `MediaSource APIs` are only supported on chrome canary which has unreliable RTP (RTCDataChannel) streams.
 3. Streaming `WebM` files only (in the moment!)
 4. WebM file's size must be less than `1000KB`; otherwise it will fail. It is a bug will be fixed soon.
 
+=
+
 #### How to stream your own video?
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/streamer.js"> </script>
+<script src="https://www.webrtc-experiment.com/streamer.js"> </script>
 ```
 
 ```javascript
 var streamer = new Streamer();
 ```
 
+=
+
 #### /* pre-recorded media sender */
 
 ```javascript
@@ -27,6 +31,8 @@ document.querySelector('input[type=file]').onchange = function () {
 };
 ```
 
+=
+
 #### /* pre-recorded media receiver */
 
 ```javascript
@@ -39,6 +45,8 @@ function onData(data) {
 }
 ```
 
+=
+
 #### /* socket.io/websocket to push chunks */
 
 ```javascript
@@ -49,6 +57,8 @@ socket.onmessage = onData;
 socket.on('message', onData);
 ```
 
+=
+
 #### It is an early release!
 
 This experiment is an early release. In future, RTCDataChannel APIs will be used to stream pre-recorded media in realtime!
@@ -57,6 +67,8 @@ This experiment is an early release. In future, RTCDataChannel APIs will be used
 
 We are waiting `video.captureStream` implementation that is proposed for pre-recorded media streaming, unfortunately still in draft!
 
+=
+
 #### In future, to stream pre-recorded medias
 
 ```javascript
@@ -76,6 +88,8 @@ partial interface HTMLMediaElement {
 // peer.addStream ( preRecordedStream );
 ```
 
+=
+
 #### How this experiment works?
 
 1. Getting access to `WebM` video file using `File API`
@@ -84,6 +98,8 @@ partial interface HTMLMediaElement {
 4. As soon as other party receives first chunk; `MediaSource API` will start playing video without waiting for all chunks to be download!
 5. You can save/store/record those chunks in any database; because it is a typed array [Uint8Array] in text form.
 
+=
+
 #### Let's say you want to:
 
 1. Stream 5min to 7 min of video data i.e. total two minutes of video data over all sockets from first WebM file.
@@ -94,19 +110,25 @@ You can do all such things today!
 
 In simple words; you can stream part of video from first WebM file; part of video from second WebM file and so on, in realtime!
 
+=
+
 #### Spec Reference
 
 1. http://www.w3.org/TR/streamproc/
 2. https://dvcs.w3.org/hg/html-media/raw-file/tip/media-source/media-source.html
 
+=
+
 #### Browser Support
 
-[Pre-recorded media streaming](https://webrtc-experiment.appspot.com/Pre-recorded-Media-Streaming/) experiment works fine on following web-browsers:
+[Pre-recorded media streaming](https://www.webrtc-experiment.com/Pre-recorded-Media-Streaming/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |:-------------:|
 | Google Chrome | [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) |
 
+=
+
 ##### License
 
-[Pre-recorded media streaming](https://webrtc-experiment.appspot.com/Pre-recorded-Media-Streaming/) experiment is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[Pre-recorded media streaming](https://www.webrtc-experiment.com/Pre-recorded-Media-Streaming/) experiment is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/Pre-recorded-Media-Streaming/index.html b/Pre-recorded-Media-Streaming/index.html
index a1edf947..7e850242 100644
--- a/Pre-recorded-Media-Streaming/index.html
+++ b/Pre-recorded-Media-Streaming/index.html
@@ -124,13 +124,13 @@
             document.createElement('article');
             document.createElement('footer');
         </script>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/streamer.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/streamer.js"> </script>
     </head>
     
     <body>
         <article>
-            <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+            <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
             <h1>
                 Pre-recorded media streaming
@@ -141,7 +141,7 @@ <h1>
             </p>
            
             <section>
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
 
             <a href="https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/Pre-recorded-Media-Streaming/WebM-file-to-stream-as-pre-recorded-media.webm" title="Download WebM file to stream as pre-recorded media" download="WebM-file-to-stream-as-pre-recorded-media.webm"
@@ -191,7 +191,7 @@ <h1>
             <br /><br />
             <h2>How to stream your own video?</h2>
             <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/streamer.js"&gt; &lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/streamer.js"&gt; &lt;/script&gt;
 &lt;script&gt; var streamer = new Streamer(); &lt;/script&gt;
 
 <h2>/* pre-recorded media sender */</h2>
@@ -289,14 +289,14 @@ <h2 id="feedback" style="padding: .2em .4em;border-bottom: 1px solid rgb(189, 18
         </article>
         <footer>
             <p>
-                <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+                <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span> 2013 </span> »
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/README.md b/README.md
index 97cce782..724e7486 100644
--- a/README.md
+++ b/README.md
@@ -1,4 +1,4 @@
-## Realtime/Working [WebRTC Experiments](https://webrtc-experiment.appspot.com/)
+## Realtime/Working [WebRTC Experiments](https://www.webrtc-experiment.com/)
 
 1. It is a repository of uniquely experimented WebRTC demos; written by <a href="https://github.com/muaz-khan">Muaz Khan</a>!
 2. No special requirement! Just Chrome (for desktop and android) or Firefox!
@@ -10,12 +10,12 @@
 
 | Library Name        | Short Description           | Documentation | Demos |
 | ------------- |-------------|-------------|-------------|
-| `RecordRTC.js` | A library for audio/video recording | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC) | [Demos](https://webrtc-experiment.appspot.com/RecordRTC/) |
-| `AudioVideoRecorder.js` | Audio+video recording using MediaRecorder API | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/AudioVideoRecorder) | [Demos](https://webrtc-experiment.appspot.com/AudioVideoRecorder/) |
-| `RTCMultiConnection.js` | An ultimate wrapper library for `RTCWeb APIs` | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection) | [Demos](https://webrtc-experiment.appspot.com/#RTCMultiConnection) |
-| `DataChannel.js` | An ultimate wrapper library for `RTCDataChannel APIs` | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel) | [Demos](https://webrtc-experiment.appspot.com/#DataChannel) |
-| `RTCall.js` | A library for voice (i.e. audio-only) calls | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCall) | [Demos](https://webrtc-experiment.appspot.com/RTCall/) |
-| `meeting.js` | A library for audio/video conferencing | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/meeting) | [Demos](https://webrtc-experiment.appspot.com/meeting/) |
+| `RecordRTC.js` | A library for audio/video recording | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC) | [Demos](https://www.webrtc-experiment.com/RecordRTC/) |
+| `AudioVideoRecorder.js` | Audio+video recording using MediaRecorder API | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/AudioVideoRecorder) | [Demos](https://www.webrtc-experiment.com/AudioVideoRecorder/) |
+| `RTCMultiConnection.js` | An ultimate wrapper library for `RTCWeb APIs` | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection) | [Demos](https://www.webrtc-experiment.com/#RTCMultiConnection) |
+| `DataChannel.js` | An ultimate wrapper library for `RTCDataChannel APIs` | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel) | [Demos](https://www.webrtc-experiment.com/#DataChannel) |
+| `RTCall.js` | A library for voice (i.e. audio-only) calls | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCall) | [Demos](https://www.webrtc-experiment.com/RTCall/) |
+| `meeting.js` | A library for audio/video conferencing | [Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/meeting) | [Demos](https://www.webrtc-experiment.com/meeting/) |
 
 =
 
@@ -23,11 +23,11 @@
 
 | Experiment Name        | Short Description           | Source Code | Demo |
 | ------------- |-------------|-------------|-------------|
-| `Pre-recorded Media Streaming` | Stream video files in realtime; same like webcam streaming! | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pre-recorded-Media-Streaming) | [Demo](https://webrtc-experiment.appspot.com/Pre-recorded-Media-Streaming/) |
-| `Part of Screen Sharing` | Share a region of the screen; not the entire screen! | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing) | [Demo](https://webrtc-experiment.appspot.com/part-of-screen-sharing/) |
-| `Plugin-free Screen Sharing` | Share the entire screen | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharinghttps://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing) | [Demo](https://webrtc-experiment.appspot.com/Pluginfree-Screen-Sharing/) |
-| `One-Way Broadcasting` | Same like radio stations; transmit audio/video/screen streams in one-way direction. Though, it is browser-to-browser streaming! | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/webrtc-broadcasting) | [Demo](https://webrtc-experiment.appspot.com/webrtc-broadcasting/) |
-| `Audio-only Calls` | Realtime, plugin-free, voice-only calls | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/realtime-pluginfree-calls) | [Demo](https://webrtc-experiment.appspot.com/calls/) |
+| `Pre-recorded Media Streaming` | Stream video files in realtime; same like webcam streaming! | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pre-recorded-Media-Streaming) | [Demo](https://www.webrtc-experiment.com/Pre-recorded-Media-Streaming/) |
+| `Part of Screen Sharing` | Share a region of the screen; not the entire screen! | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing) | [Demo](https://www.webrtc-experiment.com/part-of-screen-sharing/) |
+| `Plugin-free Screen Sharing` | Share the entire screen | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharinghttps://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing) | [Demo](https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/) |
+| `One-Way Broadcasting` | Same like radio stations; transmit audio/video/screen streams in one-way direction. Though, it is browser-to-browser streaming! | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/webrtc-broadcasting) | [Demo](https://www.webrtc-experiment.com/webrtc-broadcasting/) |
+| `Audio-only Calls` | Realtime, plugin-free, voice-only calls | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/realtime-pluginfree-calls) | [Demo](https://www.webrtc-experiment.com/calls/) |
 
 =
 
@@ -35,9 +35,9 @@
 
 | Experiment Name        | Previous Demos           | New Demos |
 | ------------- |-------------|-------------|
-| **video-conferencing** / multi-user (group) video sharing | [Demo](https://webrtc-experiment.appspot.com/video-conferencing/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing) | [Demo](https://webrtc-experiment.appspot.com/meeting/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/meeting) |
-| **file sharing** / multi-user (group) files hangout | [Demo](https://webrtc-experiment.appspot.com/file-hangout/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-hangout) | [Demo](https://webrtc-experiment.appspot.com/file-sharing/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-sharing) |
-| **text chat** / multi-user (group) text chat | [Demo](https://webrtc-experiment.appspot.com/chat-hangout/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/chat-hangout) | [Demo](https://webrtc-experiment.appspot.com/text-chat/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/text-chat) |
+| **video-conferencing** / multi-user (group) video sharing | [Demo](https://www.webrtc-experiment.com/video-conferencing/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing) | [Demo](https://www.webrtc-experiment.com/meeting/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/meeting) |
+| **file sharing** / multi-user (group) files hangout | [Demo](https://www.webrtc-experiment.com/file-hangout/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-hangout) | [Demo](https://www.webrtc-experiment.com/file-sharing/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-sharing) |
+| **text chat** / multi-user (group) text chat | [Demo](https://www.webrtc-experiment.com/chat-hangout/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/chat-hangout) | [Demo](https://www.webrtc-experiment.com/text-chat/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/text-chat) |
 
 =
 
@@ -45,8 +45,8 @@
 
 | Experiment Name        | Previous Demos           | New Demos |
 | ------------- |-------------|-------------|
-| **video-broadcasting** | [Demo](https://webrtc-experiment.appspot.com/broadcast/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/broadcast) | [Demo](https://webrtc-experiment.appspot.com/one-to-many-video-broadcasting/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/one-to-many-video-broadcasting) |
-| **audio-broadcasting** | [Demo](https://webrtc-experiment.appspot.com/audio-broadcast/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/audio-broadcast) | [Demo](https://webrtc-experiment.appspot.com/one-to-many-audio-broadcasting/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/one-to-many-audio-broadcasting) |
+| **video-broadcasting** | [Demo](https://www.webrtc-experiment.com/broadcast/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/broadcast) | [Demo](https://www.webrtc-experiment.com/one-to-many-video-broadcasting/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/one-to-many-video-broadcasting) |
+| **audio-broadcasting** | [Demo](https://www.webrtc-experiment.com/audio-broadcast/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/audio-broadcast) | [Demo](https://www.webrtc-experiment.com/one-to-many-audio-broadcasting/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/one-to-many-audio-broadcasting) |
 
 =
 
@@ -54,8 +54,8 @@
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| **One-to-one WebRTC video chat using WebSocket** | [Demo](https://webrtc-experiment.appspot.com/websocket/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/websocket) |
-| **One-to-one WebRTC video chat using socket.io** | [Demo](https://webrtc-experiment.appspot.com/socket.io/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socket.io) |
+| **One-to-one WebRTC video chat using WebSocket** | [Demo](https://www.webrtc-experiment.com/websocket/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/websocket) |
+| **One-to-one WebRTC video chat using socket.io** | [Demo](https://www.webrtc-experiment.com/socket.io/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socket.io) |
 
 =
 
@@ -63,11 +63,11 @@
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| **Share screen and audio/video from single peer connection!** | [Demo](https://webrtc-experiment.appspot.com/demos/screen-and-video-from-single-peer.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/screen-and-video-from-single-peer.html) |
-| **Text chat using RTCDataChannel APIs** | [Demo](https://webrtc-experiment.appspot.com/demos/client-side-datachannel.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/) |client-side-datachannel.html
-| **Simple video chat** | [Demo](https://webrtc-experiment.appspot.com/demos/client-side.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/client-side.html) |
-| **Sharing video - using socket.io for signaling** |[Demo](https://webrtc-experiment.appspot.com/demos/client-side-socket-io.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/client-side-socket-io.html) |
-| **Sharing video - using WebSockets for signaling** | [Demo](https://webrtc-experiment.appspot.com/demos/client-side-websocket.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/client-side-websocket.html) |
+| **Share screen and audio/video from single peer connection!** | [Demo](https://www.webrtc-experiment.com/demos/screen-and-video-from-single-peer.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/screen-and-video-from-single-peer.html) |
+| **Text chat using RTCDataChannel APIs** | [Demo](https://www.webrtc-experiment.com/demos/client-side-datachannel.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/) |client-side-datachannel.html
+| **Simple video chat** | [Demo](https://www.webrtc-experiment.com/demos/client-side.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/client-side.html) |
+| **Sharing video - using socket.io for signaling** |[Demo](https://www.webrtc-experiment.com/demos/client-side-socket-io.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/client-side-socket-io.html) |
+| **Sharing video - using WebSockets for signaling** | [Demo](https://www.webrtc-experiment.com/demos/client-side-websocket.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/client-side-websocket.html) |
 | **Audio Only Streaming** | ---- | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos/audio-only-streaming.html) |
 
 =
@@ -76,8 +76,8 @@
 
 | Experiment Name        | Previous Demos           | New Demos |
 | ------------- |-------------|-------------|
-| **Plugin-free screen sharing** / share the entire screen | [Demo](https://webrtc-experiment.appspot.com/Pluginfree-Screen-Sharing/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing) | [Demo](https://webrtc-experiment.appspot.com/screen-sharing/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/screen-sharing) |
-| **Tab sharing** / using `tabCapture` APIs | [Demo](https://webrtc-experiment.appspot.com/screen-broadcast/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/screen-broadcast) | ---- |
+| **Plugin-free screen sharing** / share the entire screen | [Demo](https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing) | [Demo](https://www.webrtc-experiment.com/screen-sharing/) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/screen-sharing) |
+| **Tab sharing** / using `tabCapture` APIs | [Demo](https://www.webrtc-experiment.com/screen-broadcast/) / [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/screen-broadcast) | ---- |
 
 =
 
@@ -85,8 +85,8 @@
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| Share part-of-screen **using RTCDataChannel APIs** | [Demo](https://webrtc-experiment.appspot.com/part-of-screen-sharing/webrtc-data-channel/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing/webrtc-and-part-of-screen-sharing) |
-| Share part-of-screen **using Firebase** | [Demo](https://webrtc-experiment.appspot.com/part-of-screen-sharing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing) |
+| Share part-of-screen **using RTCDataChannel APIs** | [Demo](https://www.webrtc-experiment.com/part-of-screen-sharing/webrtc-data-channel/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing/webrtc-and-part-of-screen-sharing) |
+| Share part-of-screen **using Firebase** | [Demo](https://www.webrtc-experiment.com/part-of-screen-sharing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing) |
 | **A realtime chat using RTCDataChannel** | [Demo](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing/realtime-chat/RTCDataChannel) |
 | **A realtime chat using Firebase** | [Demo](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/No-WebRTC-Chat.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing/realtime-chat) |
 
@@ -96,8 +96,9 @@
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| **DataChannel basic demo** | [Demo](https://webrtc-experiment.appspot.com/DataChannel/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/DataChannel/index.html) |
-| **Auto Session Establishment** | [Demo](https://webrtc-experiment.appspot.com/DataChannel/auto-session-establishment/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/DataChannel/auto-session-establishment.html) |
+| **DataChannel basic demo** | [Demo](https://www.webrtc-experiment.com/DataChannel/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/DataChannel/index.html) |
+| **Auto Session Establishment** | [Demo](https://www.webrtc-experiment.com/DataChannel/auto-session-establishment/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/DataChannel/auto-session-establishment.html) |
+| Share part-of-screen **using DataChannel.js** | [Demo](https://www.webrtc-experiment.com/part-of-screen-sharing/webrtc-data-channel/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing/webrtc-and-part-of-screen-sharing) |
 | **Private Chat** | [Demo](https://muazkh.appspot.com/privatechat/) | ---- |
 
 =
@@ -106,9 +107,9 @@
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| **Attaching Remote Media Streams** | [Demo](https://webrtc-experiment.appspot.com/remote-media-stream-attachment/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/experimental/remote-media-stream-attachment) |
-| **mozCaptureStreamUntilEnded for pre-recorded media streaming** | [Demo](https://webrtc-experiment.appspot.com/mozCaptureStreamUntilEnded/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/experimental/mozCaptureStreamUntilEnded) |
-| **Remote audio stream recording** | [Demo](https://webrtc-experiment.appspot.com/demos/remote-stream-recording.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/experimental/remote-stream-recording.html) |
+| **Attaching Remote Media Streams** | [Demo](https://www.webrtc-experiment.com/remote-media-stream-attachment/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/experimental/remote-media-stream-attachment) |
+| **mozCaptureStreamUntilEnded for pre-recorded media streaming** | [Demo](https://www.webrtc-experiment.com/mozCaptureStreamUntilEnded/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/experimental/mozCaptureStreamUntilEnded) |
+| **Remote audio stream recording** | [Demo](https://www.webrtc-experiment.com/demos/remote-stream-recording.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/experimental/remote-stream-recording.html) |
 
 =
 
@@ -116,20 +117,20 @@
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| **All-in-One test** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html) |
-| **Video Conferencing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/video-conferencing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html) |
-| **Multi-Session Establishment** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/multi-session-establishment/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html) |
-| **RTCMultiConnection-v1.3 testing demo** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html) |
-| **Video Broadcasting** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/video-broadcasting/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html) |
-| **File Sharing + Text Chat** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/group-file-sharing-plus-text-chat/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html) |
-| **Audio Conferencing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/audio-conferencing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html) |
-| **Join with/without camera** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/join-with-or-without-camera/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html) |
-| **Screen Sharing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/screen-sharing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html) |
-| **One-to-One file sharing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/one-to-one-filesharing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html) |
-| **Manual session establishment + extra data transmission** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.htmlhttps://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html) |
-| **Manual session establishment + extra data transmission + video conferencing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) |
-| **Customizing Bandwidth** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/bandwidth/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html) |
-| **Users ejection and presence detection** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/users-ejection/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html) |
+| **All-in-One test** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html) |
+| **Video Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/videoconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html) |
+| **Multi-Session Establishment** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/multi-session-establishment.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html) |
+| **RTCMultiConnection-v1.3 testing demo** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/RTCMultiConnection-v1.3-demo.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html) |
+| **Video Broadcasting** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/video-broadcasting.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html) |
+| **File Sharing + Text Chat** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/group-file-sharing-plus-text-chat/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html) |
+| **Audio Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/audioconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html) |
+| **Join with/without camera** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/join-with-or-without-camera.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html) |
+| **Screen Sharing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/screen-sharing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html) |
+| **One-to-One file sharing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/one-to-one-filesharing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html) |
+| **Manual session establishment + extra data transmission** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.htmlhttps://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html) |
+| **Manual session establishment + extra data transmission + video conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) |
+| **Customizing Bandwidth** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/bandwidth.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html) |
+| **Users ejection and presence detection** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/users-ejection.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html) |
 | **RTCMultiConnection-v1.3 and socket.io** | ---- | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-and-socket.io.html) |
 
 =
@@ -138,10 +139,10 @@
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| **All-in-One test** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4-Demos/All-in-One.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html) |
-| **Renegotiation & Mute/UnMute/Stop** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4-Demos/Renegotiation.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Renegotiation.html) |
-| **Video-Conferencing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html) |
-| **Multi-streams attachment** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html) |
+| **All-in-One test** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4-Demos/All-in-One.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html) |
+| **Renegotiation & Mute/UnMute/Stop** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4-Demos/Renegotiation.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Renegotiation.html) |
+| **Video-Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html) |
+| **Multi-streams attachment** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html) |
 
 =
 
@@ -152,18 +153,18 @@
 | [RTCMultiConnection Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection) |
 | [DataChannel Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel) |
 | [RTCPeerConnection Documentation](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCPeerConnection) |
-| [How to use RTCPeerConnection.js?](https://webrtc-experiment.appspot.com/docs/how-to-use-rtcpeerconnection-js-v1.1.html) |
-| [RTCDataChannel for Beginners](https://webrtc-experiment.appspot.com/docs/rtc-datachannel-for-beginners.html) |
-| [How to use RTCDataChannel?](https://webrtc-experiment.appspot.com/docs/how-to-use-rtcdatachannel.html) - single code for both canary and nightly |
-| [WebRTC for Beginners: A getting stared guide!](https://webrtc-experiment.appspot.com/docs/webrtc-for-beginners.html) |
-| [WebRTC for Newbies ](https://webrtc-experiment.appspot.com/docs/webrtc-for-newbies.html) |
+| [How to use RTCPeerConnection.js?](https://www.webrtc-experiment.com/docs/how-to-use-rtcpeerconnection-js-v1.1.html) |
+| [RTCDataChannel for Beginners](https://www.webrtc-experiment.com/docs/rtc-datachannel-for-beginners.html) |
+| [How to use RTCDataChannel?](https://www.webrtc-experiment.com/docs/how-to-use-rtcdatachannel.html) - single code for both canary and nightly |
+| [WebRTC for Beginners: A getting stared guide!](https://www.webrtc-experiment.com/docs/webrtc-for-beginners.html) |
+| [WebRTC for Newbies ](https://www.webrtc-experiment.com/docs/webrtc-for-newbies.html) |
 
 =
 
 ##### [DataChannel.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel) / A library for RTCDataChannel APIs
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/DataChannel.js"></script>
+<script src="https://www.webrtc-experiment.com/DataChannel.js"></script>
 ```
 
 ```javascript
@@ -185,51 +186,36 @@ channel.channels['muazkh'].send(file || data || 'text');
 ##### [RTCMultiConnection.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection) / A library for all WebRTC APIs
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"></script>
-```
+<script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"> </script>
 
-```javascript
-var connection = new RTCMultiConnection();
-
-// session is included in v1.3 and higher releases
-connection.session = {
-    audio: true,
-    video: true,
-    screen: true,
-    oneway: true
-};
+<button id="init">Open New Connection</button><br />
 
-// bandwidth is included in v1.3 and higher releases
-connection.bandwidth = {
-    audio: 50,
-    video: 256,
-    data: 1638400
-};
+<script>
+    var connection = new RTCMultiConnection();
 
-// framerate is included in v1.4 and higher releases
-connection.framerate = {
-    minptime: 10,
-    maxptime: 60
-};
+    connection.session = {
+        audio: true,
+        video: true
+    };
 
-connection.onstream = function (e) {
-    if (e.type === 'local') mainVideo.src = e.blobURL; // or URL.createObjectURL(e.stream)
-    if (e.type === 'remote') document.body.appendChild(e.mediaElement);
-}
+    connection.onstream = function(e) {
+        document.body.appendChild(e.mediaElement);
+    };
 
-// searching/connecting pre-created session
-connection.connect('session-id');
+    connection.onstreamended = function(e) {
+        if (e.mediaElement.parentNode) e.mediaElement.parentNode.removeChild(e.mediaElement);
+    };
 
-// to create/open a new session
-[init-button].onclick = function() {
-    connection.open('session-id');
-};
+    var session_unique_idetifier = 'Session Unique Identifier';
 
-// runtime/dynamic streams insertion i.e. renegotiation
-connection.peers['user-id'].addStream({
-    audio: true,
-    video: true
-});
+    connection.connect(session_unique_idetifier);
+
+    document.getElementById('init').onclick = function() {
+        this.disabled = true;
+        connection.open(session_unique_idetifier);
+    };
+</script>
 ```
 
 =
@@ -237,7 +223,7 @@ connection.peers['user-id'].addStream({
 ##### [RTCall.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCall) / A library for audio-only calling
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RTCall.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCall.js"></script>
 ```
 
 ```javascript
@@ -311,15 +297,15 @@ Want to use XHR, WebSockets, SIP, XMPP, etc. for signaling? Read [this post](htt
 
 =
 
-##### How to use [RecordRTC](https://webrtc-experiment.appspot.com/RecordRTC/)?
+##### How to use [RecordRTC](https://www.webrtc-experiment.com/RecordRTC/)?
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RecordRTC.js"></script>
+<script src="https://www.webrtc-experiment.com/RecordRTC.js"></script>
 ```
 
 =
 
-##### How to record video using [RecordRTC](https://webrtc-experiment.appspot.com/RecordRTC/)?
+##### How to record video using [RecordRTC](https://www.webrtc-experiment.com/RecordRTC/)?
 
 ```javascript
 var recorder = RecordRTC({
@@ -337,10 +323,10 @@ recorder.stopVideo(function(recordedFileURL) {
 
 =
 
-##### How to record GIF using [RecordRTC](https://webrtc-experiment.appspot.com/RecordRTC/)?
+##### How to record GIF using [RecordRTC](https://www.webrtc-experiment.com/RecordRTC/)?
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/gif-recorder.js"></script>
+<script src="https://www.webrtc-experiment.com/gif-recorder.js"></script>
 ```
 
 ```javascript
@@ -359,7 +345,7 @@ recorder.stopGIF(function(gifURL) {
 
 =
 
-##### How to record audio using [RecordRTC](https://webrtc-experiment.appspot.com/RecordRTC/)?
+##### How to record audio using [RecordRTC](https://www.webrtc-experiment.com/RecordRTC/)?
 
 ```javascript
 var recorder = RecordRTC({
@@ -381,7 +367,7 @@ recorder.stopAudio(function(recordedFileURL) {
 
 =
 
-##### How to record audio using [AudioVideoRecorder](https://webrtc-experiment.appspot.com/AudioVideoRecorder/)?
+##### How to record audio using [AudioVideoRecorder](https://www.webrtc-experiment.com/AudioVideoRecorder/)?
 
 ```javascript
 AudioVideoRecorder({
@@ -418,7 +404,7 @@ AudioVideoRecorder({
 
 ##### Browser Support
 
-[WebRTC Experiments](https://webrtc-experiment.appspot.com) works fine on following web-browsers:
+[WebRTC Experiments](https://www.webrtc-experiment.com/) works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -430,4 +416,4 @@ AudioVideoRecorder({
 
 ##### License
 
-[WebRTC Experiments](https://webrtc-experiment.appspot.com/) are released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC Experiments](https://www.webrtc-experiment.com/) are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/RTCDataConnection/README.md b/RTCDataConnection/README.md
index ef0ee0d0..62a2b1ae 100644
--- a/RTCDataConnection/README.md
+++ b/RTCDataConnection/README.md
@@ -37,14 +37,19 @@ rtcDataConnection.initDataConnection();
 rtcDataConnection.send( file || data || 'text' );
 ```
 
+=
+
 #### Features
 
 1. Share files of any size � directly
 2. Share text messages of any length
 3. Share any kind of data � directly
 
+=
+
 #### Read [RTCDataConnection Documentation](http://bit.ly/RTCDataConnection)
 
+=
 
 #### Browser Support
 [RTCDataConnection](http://bit.ly/RTCDataConnection) works fine on following web-browsers:
@@ -53,9 +58,9 @@ rtcDataConnection.send( file || data || 'text' );
 | ------------- |:-------------:|
 | Firefox | [Stable](http://www.mozilla.org/en-US/firefox/new/) / [Aurora](http://www.mozilla.org/en-US/firefox/aurora/) / [Nightly](http://nightly.mozilla.org/) |
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
-| Internet Explorer / IE | [Chrome Frame](http://www.google.com/chromeframe) |
 
+=
 
 #### License
 
-[RTCDataConnection](http://bit.ly/RTCDataConnection) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[RTCDataConnection](http://bit.ly/RTCDataConnection) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/RTCDataConnection/RTCDataConnection-Manual-Test.html b/RTCDataConnection/RTCDataConnection-Manual-Test.html
index e8a691df..711bddf3 100644
--- a/RTCDataConnection/RTCDataConnection-Manual-Test.html
+++ b/RTCDataConnection/RTCDataConnection-Manual-Test.html
@@ -149,7 +149,7 @@ <h1>
 				Copyright © 2013 <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a>&lt;<a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>&gt;.	
 			</p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             
 			<div id="result"></div>
@@ -269,7 +269,7 @@ <h2>How to use RTCDataConnection? <a href="https://bit.ly/RTCDataConnection" tar
         </article>
         <footer>
             <p>
-                <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+                <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span> 2013 </span> »
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
diff --git a/RTCDataConnection/RTCDataConnection-Simple-Test.html b/RTCDataConnection/RTCDataConnection-Simple-Test.html
index 7166dcdd..25c21368 100644
--- a/RTCDataConnection/RTCDataConnection-Simple-Test.html
+++ b/RTCDataConnection/RTCDataConnection-Simple-Test.html
@@ -150,7 +150,7 @@ <h1>
 			</p>
             
 			<section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             
 			<div id="result"></div>
@@ -247,7 +247,7 @@ <h2>How to use RTCDataConnection? <a href="https://bit.ly/RTCDataConnection" tar
         </article>
         <footer>
             <p>
-                <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+                <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span> 2013 </span> »
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
diff --git a/RTCDataConnection/RTCDataConnection-Test.html b/RTCDataConnection/RTCDataConnection-Test.html
index 96ee555f..da188f6b 100644
--- a/RTCDataConnection/RTCDataConnection-Test.html
+++ b/RTCDataConnection/RTCDataConnection-Test.html
@@ -149,7 +149,7 @@ <h1>
 				Copyright © 2013 <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a>&lt;<a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>&gt;.	
 			</p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             
 			<div id="result"></div>
@@ -249,7 +249,7 @@ <h2>How to use RTCDataConnection? <a href="https://bit.ly/RTCDataConnection" tar
         </article>
         <footer>
             <p>
-                <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+                <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span> 2013 </span> »
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
diff --git a/RTCMultiConnection/README.md b/RTCMultiConnection/README.md
index 8f6b794c..c8b6a32c 100644
--- a/RTCMultiConnection/README.md
+++ b/RTCMultiConnection/README.md
@@ -1,4 +1,4 @@
-# [RTCMultiConnection-v1.4](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js) Documentation
+# [RTCMultiConnection-v1.4](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js) Documentation
 
 ##### Features
 
@@ -19,8 +19,8 @@ and much more!
 ##### A Quick Demo using Firebase for Signaling
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"> </script>
+<script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"> </script>
 
 <style>
     video {
@@ -63,8 +63,8 @@ and much more!
 ##### A Quick Demo using socket.io for Signaling
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/socket.io.js"> </script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"> </script>
+<script src="https://www.webrtc-experiment.com/socket.io.js"> </script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"> </script>
 
 <style>
     video {
@@ -139,7 +139,7 @@ and much more!
 ##### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"></script>
 ```
 
 ##### Second Step: Start using it!
@@ -714,24 +714,24 @@ io.sockets.on('connection', function (socket) {
 
 =
 
-##### [RTCMultiConnection Demos](https://webrtc-experiment.appspot.com/#RTCMultiConnection)
+##### [RTCMultiConnection Demos](https://www.webrtc-experiment.com/#RTCMultiConnection)
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| **All-in-One test** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html) |
-| **Video Conferencing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/video-conferencing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html) |
-| **Multi-Session Establishment** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/multi-session-establishment/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html) |
-| **RTCMultiConnection-v1.3 testing demo** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html) |
-| **Video Broadcasting** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/video-broadcasting/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html) |
-| **File Sharing + Text Chat** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/group-file-sharing-plus-text-chat/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html) |
-| **Audio Conferencing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/audio-conferencing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html) |
-| **Join with/without camera** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/join-with-or-without-camera/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html) |
-| **Screen Sharing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/screen-sharing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html) |
-| **One-to-One file sharing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/one-to-one-filesharing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html) |
-| **Manual session establishment + extra data transmission** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.htmlhttps://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html) |
-| **Manual session establishment + extra data transmission + video conferencing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) |
-| **Customizing Bandwidth** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/bandwidth/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html) |
-| **Users ejection and presence detection** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection/users-ejection/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html) |
+| **All-in-One test** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html) |
+| **Video Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/videoconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html) |
+| **Multi-Session Establishment** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/multi-session-establishment.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html) |
+| **RTCMultiConnection-v1.3 testing demo** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/RTCMultiConnection-v1.3-demo.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html) |
+| **Video Broadcasting** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/video-broadcasting.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html) |
+| **File Sharing + Text Chat** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/group-file-sharing-plus-text-chat/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html) |
+| **Audio Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/audioconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html) |
+| **Join with/without camera** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/join-with-or-without-camera.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html) |
+| **Screen Sharing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/screen-sharing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html) |
+| **One-to-One file sharing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/one-to-one-filesharing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html) |
+| **Manual session establishment + extra data transmission** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.htmlhttps://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html) |
+| **Manual session establishment + extra data transmission + video conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) |
+| **Customizing Bandwidth** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/bandwidth.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html) |
+| **Users ejection and presence detection** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/users-ejection.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html) |
 | **RTCMultiConnection-v1.3 and socket.io** | ---- | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-and-socket.io.html) |
 
 =
@@ -740,10 +740,10 @@ io.sockets.on('connection', function (socket) {
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| **All-in-One test** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4-Demos/All-in-One.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html) |
-| **Renegotiation & Mute/UnMute/Stop** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4-Demos/Renegotiation.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Renegotiation.html) |
-| **Video-Conferencing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html) |
-| **Multi-streams attachment** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html) |
+| **All-in-One test** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4-Demos/All-in-One.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html) |
+| **Renegotiation & Mute/UnMute/Stop** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4-Demos/Renegotiation.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Renegotiation.html) |
+| **Video-Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html) |
+| **Multi-streams attachment** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html) |
 
 =
 
@@ -758,7 +758,7 @@ io.sockets.on('connection', function (socket) {
 
 ##### Browser Support
 
-[RTCMultiConnection-v1.4.js](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js) supports following browsers:
+[RTCMultiConnection-v1.4.js](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js) supports following browsers:
 
 | Browser        | Support           |
 | ------------- |:-------------|
@@ -770,4 +770,4 @@ io.sockets.on('connection', function (socket) {
 
 ##### License
 
-[RTCMultiConnection-v1.4.js](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[RTCMultiConnection-v1.4.js](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-and-socket.io.html b/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-and-socket.io.html
index 9788a37f..b8dc307d 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-and-socket.io.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-and-socket.io.html
@@ -1,5 +1,5 @@
-<script src="https://webrtc-experiment.appspot.com/socket.io.js"></script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js"></script>
+<script src="https://www.webrtc-experiment.com/socket.io.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js"></script>
 
 <button id="init">Open NEW Session</button>
 
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html b/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html
index 70d7ad34..a82eef56 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html
@@ -16,8 +16,8 @@
                 location.reload();
             }
         </script>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js"></script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js"></script>
     </head>
     
     <body>
@@ -29,7 +29,7 @@
         <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://www.webrtc-experiment.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>
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html b/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html
index 9aeeabd3..9adc3d8f 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html
@@ -202,12 +202,12 @@
             document.createElement('article');
             document.createElement('footer');
         </script>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js"></script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"></script>
     </head>
     
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none; right: 0; top: 0;">↑ WEBRTC EXPERIMENTS</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none; right: 0; top: 0;">↑ WEBRTC EXPERIMENTS</a>
 
              <h1>
             <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">RTCMultiConnection</a>
@@ -219,7 +219,7 @@ <h1>
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <section>
                  <h2>Open New Session:</h2>
@@ -400,7 +400,7 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
              <h2>Getting started with <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
             target="_blank">RTCMultiConnection</a></h2>
  <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var connection = new <strong>RTCMultiConnection</strong>();
     connection.<strong>open</strong>('session-id');    // open a new session
@@ -450,13 +450,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                 href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+        <script src="https://www.webrtc-experiment.com/common.js"></script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/audio-conferencing-data-sharing.html b/RTCMultiConnection/RTCMultiConnection-Demos/audio-conferencing-data-sharing.html
index 114ced99..13c11282 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/audio-conferencing-data-sharing.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/audio-conferencing-data-sharing.html
@@ -169,13 +169,13 @@
             document.createElement('footer');
         </script>
 
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-		<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js"></script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+		<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"></script>
     </head>
 
     <body>
         <article>
-            <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑ WEBRTC EXPERIMENTS</a>
+            <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑ WEBRTC EXPERIMENTS</a>
 
 
             <h1>
@@ -188,7 +188,7 @@ <h1>
                                                   Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
 
             <section>
@@ -338,7 +338,7 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
             <h2>Getting started with <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
                                         target="_blank">RTCMultiConnection</a></h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var room = new <strong>RTCMultiConnection</strong>('session-id');
     room.session = 'audio + data';
@@ -358,7 +358,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         </article>
         <footer>
             <p>
-                <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+                <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
                 <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -367,5 +367,5 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             </p>
         </footer>
     </body>
-	<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+	<script src="https://www.webrtc-experiment.com/common.js"></script>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html b/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html
index dca17420..f01e9a98 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html
@@ -170,12 +170,12 @@
             document.createElement('article');
             document.createElement('footer');
         </script>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js"></script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"></script>
     </head>
     
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
              <h1>
         audio-conferencing using <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
@@ -187,7 +187,7 @@ <h1>
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <section>
                  <h2>Join a session:</h2>
@@ -258,7 +258,7 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
              <h2>Getting started with <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
                                 target="_blank">RTCMultiConnection</a></h2>
  <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var connection = new <strong>RTCMultiConnection</strong>();
     connection.session = 'only audio';
@@ -304,13 +304,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                 href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+        <script src="https://www.webrtc-experiment.com/common.js"></script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html b/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html
index fbbecba8..cfd357ee 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html
@@ -187,7 +187,7 @@
             .eject {
                 background-position: center center !important;
                 background-repeat: no-repeat !important;
-                background-image: url('https://webrtc-experiment.appspot.com/images/eject.png') !important;
+                background-image: url('https://www.webrtc-experiment.com/images/eject.png') !important;
                 width: 30px;
                 height: 30px;
                 position: absolute;
@@ -202,13 +202,13 @@
             document.createElement('article');
             document.createElement('footer');
         </script>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js"></script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js"></script>
     </head>
     
     <body>
         <article>
-<a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
+<a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
     WEBRTC EXPERIMENTS</a>
 
             
@@ -219,7 +219,7 @@ <h1>Customizing Bandwidth using <a href="https://github.com/muaz-khan/WebRTC-Exp
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <section>
                  <h2>New Session:</h2>
@@ -397,13 +397,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                 href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+        <script src="https://www.webrtc-experiment.com/common.js"></script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html b/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html
index feddb870..cc317172 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html
@@ -183,13 +183,13 @@
         document.createElement('footer');
     </script>
 
-    <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js"></script>
+    <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+    <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js"></script>
 </head>
 
 <body>
 <article>
-<a href="https://webrtc-experiment.appspot.com/"
+<a href="https://www.webrtc-experiment.com/"
    style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
     WEBRTC EXPERIMENTS</a>
 
@@ -204,7 +204,7 @@ <h1>File Sharing + Text Chat using <a
         href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
 </p>
 <section class="plusone-gplus">
-    <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+    <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
 </section>
 <section>
     <h2>Open Data Channel</h2>
@@ -309,7 +309,7 @@ <h2 style="display: block; font-size: 1em; text-align: center;">Share Files</h2>
 <h2>Getting started with <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
                             target="_blank">RTCMultiConnection</a></h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var connection = new <strong>RTCMultiConnection</strong>();
 	
@@ -405,7 +405,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -413,6 +413,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html b/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html
index f80cb6a5..6a4e94e9 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html
@@ -218,13 +218,13 @@
     document.createElement('footer');
 </script>
 
-<script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js"></script>
+<script src="https://www.webrtc-experiment.com/firebase.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js"></script>
 </head>
 
 <body>
 <article>
-    <a href="https://webrtc-experiment.appspot.com/"
+    <a href="https://www.webrtc-experiment.com/"
        style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
         WEBRTC EXPERIMENTS</a>
 
@@ -240,7 +240,7 @@ <h1>
         Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
     </p>
     <section class="plusone-gplus">
-        <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+        <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
     </section>
 
     <section>
@@ -324,7 +324,7 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
     <h2>Getting started with <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
                                 target="_blank">RTCMultiConnection</a></h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var connection = new <strong>RTCMultiConnection</strong>();
 
@@ -373,7 +373,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -381,6 +381,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html b/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html
index 53633777..f73263ea 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html
@@ -203,13 +203,13 @@
     document.createElement('footer');
 </script>
 
-<script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js"></script>
+<script src="https://www.webrtc-experiment.com/firebase.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"></script>
 </head>
 
 <body>
 <article>
-    <a href="https://webrtc-experiment.appspot.com/"
+    <a href="https://www.webrtc-experiment.com/"
        style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
         WEBRTC EXPERIMENTS</a>
 
@@ -222,7 +222,7 @@ <h1>Manual session establishment + extra data transmission + video conferencing<
             href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
     </p>
     <section class="plusone-gplus">
-        <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+        <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
     </section>
     <section>
         <h2>New Session:</h2>
@@ -408,7 +408,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -416,6 +416,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html b/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html
index 43866dff..08c257fd 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html
@@ -183,13 +183,13 @@
         document.createElement('footer');
     </script>
 
-    <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js"></script>
+    <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+    <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"></script>
 </head>
 
 <body>
 <article>
-<a href="https://webrtc-experiment.appspot.com/"
+<a href="https://www.webrtc-experiment.com/"
    style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
     WEBRTC EXPERIMENTS</a>
 
@@ -203,7 +203,7 @@ <h1>Manual session establishment + extra data transmission
         href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
 </p>
 <section class="plusone-gplus">
-    <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+    <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
 </section>
 <section>
     <h2>New Session:</h2>
@@ -405,7 +405,7 @@ <h2 id="go">Go ahead and try!</h2>
 <h2><a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
        target="_blank">RTCMultiConnection</a> v1.2</h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js</a>"&gt;&lt;/script&gt;
 </pre>
 
 <br/>
@@ -423,7 +423,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -431,6 +431,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html b/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html
index 1f6098ca..686ea4fb 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html
@@ -179,13 +179,13 @@
         document.createElement('article');
         document.createElement('footer');
     </script>
-    <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js"></script>
+    <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+    <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js"></script>
 </head>
 
 <body>
 <article>
-<a href="https://webrtc-experiment.appspot.com/"
+<a href="https://www.webrtc-experiment.com/"
    style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
     WEBRTC EXPERIMENTS</a>
 
@@ -230,7 +230,7 @@ <h2>Join one of following sessions</h2>
     </li>
 </ol>
 <section>
-    <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+    <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
 </section>
 <section>
     <h2>Open Five Unique Sessions</h2>
@@ -373,7 +373,7 @@ <h2>How to open multi-sessions using <a
         href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">RTCMultiConnection</a>?
 </h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var connection = new RTCMultiConnection();
 
@@ -486,7 +486,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -494,7 +494,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html b/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html
index 121c2ce7..e58cbe01 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html
@@ -183,13 +183,13 @@
         document.createElement('footer');
     </script>
 
-    <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js"></script>
+    <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+    <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js"></script>
 </head>
 
 <body>
 <article>
-<a href="https://webrtc-experiment.appspot.com/"
+<a href="https://www.webrtc-experiment.com/"
    style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
     WEBRTC EXPERIMENTS</a>
 
@@ -204,7 +204,7 @@ <h1>One-to-One File Sharing using <a
         href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
 </p>
 <section class="plusone-gplus">
-    <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+    <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
 </section>
 <section>
     <h2>Open Data Channel</h2>
@@ -290,7 +290,7 @@ <h2 style="display: block; font-size: 1em; text-align: center;">Share Files</h2>
 <h2>Getting started with <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
                             target="_blank">RTCMultiConnection</a></h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var connection = new <strong>RTCMultiConnection</strong>();
 
@@ -416,7 +416,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -424,6 +424,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html b/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html
index 47becaee..63f81f7f 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html
@@ -218,13 +218,13 @@
     document.createElement('footer');
 </script>
 
-<script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js"></script>
+<script src="https://www.webrtc-experiment.com/firebase.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js"></script>
 </head>
 
 <body>
 <article>
-    <a href="https://webrtc-experiment.appspot.com/"
+    <a href="https://www.webrtc-experiment.com/"
        style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
         WEBRTC EXPERIMENTS</a>
 
@@ -239,7 +239,7 @@ <h1>
         Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
     </p>
     <section class="plusone-gplus">
-        <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+        <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
     </section>
 
     <section>
@@ -319,7 +319,7 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
     <h2>Getting started with <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
                                 target="_blank">RTCMultiConnection</a></h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.1.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var connection = new <strong>RTCMultiConnection</strong>();
 
@@ -371,7 +371,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -379,6 +379,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html b/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html
index ab156241..5d271314 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html
@@ -198,7 +198,7 @@
     .eject {
         background-position: center center !important;
         background-repeat: no-repeat !important;
-        background-image: url('https://webrtc-experiment.appspot.com/images/eject.png') !important;
+        background-image: url('https://www.webrtc-experiment.com/images/eject.png') !important;
         width: 30px;
         height: 30px;
         position: absolute;
@@ -214,13 +214,13 @@
     document.createElement('footer');
 </script>
 
-<script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js"></script>
+<script src="https://www.webrtc-experiment.com/firebase.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"></script>
 </head>
 
 <body>
 <article>
-<a href="https://webrtc-experiment.appspot.com/"
+<a href="https://www.webrtc-experiment.com/"
    style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
     WEBRTC EXPERIMENTS</a>
 
@@ -233,7 +233,7 @@ <h1>Users ejection and presence detection</h1>
         href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
 </p>
 <section class="plusone-gplus">
-    <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+    <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
 </section>
 <section>
     <h2>New Session:</h2>
@@ -470,7 +470,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -478,6 +478,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html b/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html
index 455af21a..9c921b4c 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html
@@ -218,13 +218,13 @@
     document.createElement('footer');
 </script>
 
-<script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"></script>
+<script src="https://www.webrtc-experiment.com/firebase.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"></script>
 </head>
 
 <body>
 <article>
-    <a href="https://webrtc-experiment.appspot.com/"
+    <a href="https://www.webrtc-experiment.com/"
        style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
         WEBRTC EXPERIMENTS</a>
 
@@ -239,7 +239,7 @@ <h1>
         Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
     </p>
     <section class="plusone-gplus">
-        <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+        <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
     </section>
 
     <section>
@@ -305,7 +305,7 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
     <h2>Getting started with <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
                                 target="_blank">RTCMultiConnection</a></h2>
             <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js</a>"&gt;&lt;/script&gt;
+&lt;script src="<a href="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js" target="_blank">https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js</a>"&gt;&lt;/script&gt;
 &lt;script&gt;
     var connection = new <strong>RTCMultiConnection</strong>();
 
@@ -354,7 +354,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -362,6 +362,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html b/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html
index d00e9ce1..b3a75188 100644
--- a/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html
+++ b/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html
@@ -95,8 +95,8 @@
 
             footer { text-align: center; }
         </style>
-		<script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js"></script>
+		<script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js"></script>
     </head>
     <body>
         <h1>
@@ -107,7 +107,7 @@ <h1>
                                               Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <section>
             <h2>Open new session:</h2>
@@ -176,12 +176,12 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                     href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.2-and-earlier.md b/RTCMultiConnection/RTCMultiConnection-v1.2-and-earlier.md
index 6e7f70ed..e52eb41b 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.2-and-earlier.md
+++ b/RTCMultiConnection/RTCMultiConnection-v1.2-and-earlier.md
@@ -1,4 +1,4 @@
-#### [RTCMultiConnection-v1.2](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js) Documentation
+#### [RTCMultiConnection-v1.2](https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js) Documentation
 
 RTCMultiConnection.js a javascript library supports features like audio/video conferencing; (one-way and one-to-many) broadcasting; screen sharing; data/text/file sharing (of any size); multi-and-manual sessions establishment; users ejection, rejection and presence detection; and more. It automatically keeps session "active" all the time; even if initiator leaves.
 
@@ -25,7 +25,7 @@ RTCMultiConnection.js a javascript library supports features like audio/video co
 #### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"></script>
 ```
 
 #### Second Step: Start using it!
@@ -573,20 +573,25 @@ There are **many other** use-cases of multi-sessions.
 
 ----
 
-#### [RTCMultiConnection Demos](https://webrtc-experiment.appspot.com/#RTCMultiConnection)
-
-1. [Multi-Session Establishment](https://webrtc-experiment.appspot.com/RTCMultiConnection/multi-session-establishment/)
-2. [File Sharing + Text Chat](https://webrtc-experiment.appspot.com/RTCMultiConnection/group-file-sharing-plus-text-chat/)
-3. [All-in-One test](https://webrtc-experiment.appspot.com/RTCMultiConnection/)
-4. [Video Conferencing](https://webrtc-experiment.appspot.com/RTCMultiConnection/video-conferencing/)
-5. [Video Broadcasting](https://webrtc-experiment.appspot.com/RTCMultiConnection/video-broadcasting/)
-6. [Audio Conferencing](https://webrtc-experiment.appspot.com/RTCMultiConnection/audio-conferencing/)
-7. [Join with/without camera](https://webrtc-experiment.appspot.com/RTCMultiConnection/join-with-or-without-camera/)
-8. [Screen Sharing](https://webrtc-experiment.appspot.com/RTCMultiConnection/screen-sharing/)
-9. [One-to-One file sharing](https://webrtc-experiment.appspot.com/RTCMultiConnection/one-to-one-filesharing/)
-10. [Manual session establishment + extra data transmission](https://webrtc-experiment.appspot.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission/)
-11. [Manual session establishment + extra data transmission + video conferencing](https://webrtc-experiment.appspot.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing/)
-12. [Users ejection and presence detection](https://webrtc-experiment.appspot.com/RTCMultiConnection/users-ejection/)
+#### [RTCMultiConnection Demos](https://www.webrtc-experiment.com/#RTCMultiConnection)
+
+| Experiment Name        | Demo           | Source Code |
+| ------------- |-------------|-------------|
+| **All-in-One test** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html) |
+| **Video Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/videoconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html) |
+| **Multi-Session Establishment** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/multi-session-establishment.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html) |
+| **RTCMultiConnection-v1.3 testing demo** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/RTCMultiConnection-v1.3-demo.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html) |
+| **Video Broadcasting** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/video-broadcasting.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html) |
+| **File Sharing + Text Chat** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/group-file-sharing-plus-text-chat/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html) |
+| **Audio Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/audioconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html) |
+| **Join with/without camera** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/join-with-or-without-camera.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html) |
+| **Screen Sharing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/screen-sharing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html) |
+| **One-to-One file sharing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/one-to-one-filesharing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html) |
+| **Manual session establishment + extra data transmission** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.htmlhttps://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html) |
+| **Manual session establishment + extra data transmission + video conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) |
+| **Customizing Bandwidth** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/bandwidth.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html) |
+| **Users ejection and presence detection** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/users-ejection.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html) |
+| **RTCMultiConnection-v1.3 and socket.io** | ---- | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-and-socket.io.html) |
 
 ----
 
@@ -606,4 +611,4 @@ if(stream.direction === RTCDirection.OneWay) {}
 
 #### License
 
-[RTCMultiConnection.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[RTCMultiConnection.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.3.md b/RTCMultiConnection/RTCMultiConnection-v1.3.md
index c3867678..25d423c9 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.3.md
+++ b/RTCMultiConnection/RTCMultiConnection-v1.3.md
@@ -1,4 +1,4 @@
-#### [RTCMultiConnection-v1.3](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js) Documentation
+#### [RTCMultiConnection-v1.3](https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js) Documentation
 
 Supports features like
 
@@ -24,7 +24,7 @@ and much more.
 #### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js"></script>
 ```
 
 #### Second Step: Start using it!
@@ -502,28 +502,31 @@ io.sockets.on('connection', function (socket) {
 
 ----
 
-#### [RTCMultiConnection Demos](https://webrtc-experiment.appspot.com/#RTCMultiConnection)
-
-1. [Multi-Session Establishment](https://webrtc-experiment.appspot.com/RTCMultiConnection/multi-session-establishment/)
-2. [File Sharing + Text Chat](https://webrtc-experiment.appspot.com/RTCMultiConnection/group-file-sharing-plus-text-chat/)
-3. [All-in-One test](https://webrtc-experiment.appspot.com/RTCMultiConnection/)
-4. [Video Conferencing](https://webrtc-experiment.appspot.com/RTCMultiConnection/video-conferencing/)
-5. [Video Broadcasting](https://webrtc-experiment.appspot.com/RTCMultiConnection/video-broadcasting/)
-6. [Audio Conferencing](https://webrtc-experiment.appspot.com/RTCMultiConnection/audio-conferencing/)
-7. [Join with/without camera](https://webrtc-experiment.appspot.com/RTCMultiConnection/join-with-or-without-camera/)
-8. [Screen Sharing](https://webrtc-experiment.appspot.com/RTCMultiConnection/screen-sharing/)
-9. [One-to-One file sharing](https://webrtc-experiment.appspot.com/RTCMultiConnection/one-to-one-filesharing/)
-10. [Manual session establishment + extra data transmission](https://webrtc-experiment.appspot.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission/)
-11. [Manual session establishment + extra data transmission + video conferencing](https://webrtc-experiment.appspot.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing/)
-12. [Users ejection and presence detection](https://webrtc-experiment.appspot.com/RTCMultiConnection/users-ejection/)
-13. [Customizing Bandwidth](https://webrtc-experiment.appspot.com/RTCMultiConnection/bandwidth/)
-14. [RTCMultiConnection-v1.3 testing demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3/)
+#### [RTCMultiConnection Demos](https://www.webrtc-experiment.com/#RTCMultiConnection)
+
+| Experiment Name        | Demo           | Source Code |
+| ------------- |-------------|-------------|
+| **All-in-One test** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html) |
+| **Video Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/videoconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/videoconferencing.html) |
+| **Multi-Session Establishment** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/multi-session-establishment.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/multi-session-establishment.html) |
+| **RTCMultiConnection-v1.3 testing demo** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/RTCMultiConnection-v1.3-demo.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-demo.html) |
+| **Video Broadcasting** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/video-broadcasting.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/video-broadcasting.html) |
+| **File Sharing + Text Chat** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/group-file-sharing-plus-text-chat/) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/group-file-sharing-plus-text-chat.html) |
+| **Audio Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/audioconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/audioconferencing.html) |
+| **Join with/without camera** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/join-with-or-without-camera.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/join-with-or-without-camera.html) |
+| **Screen Sharing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/screen-sharing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/screen-sharing.html) |
+| **One-to-One file sharing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/one-to-one-filesharing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/one-to-one-filesharing.html) |
+| **Manual session establishment + extra data transmission** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.htmlhttps://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission.html) |
+| **Manual session establishment + extra data transmission + video conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html) |
+| **Customizing Bandwidth** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/bandwidth.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/bandwidth.html) |
+| **Users ejection and presence detection** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection/users-ejection.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/users-ejection.html) |
+| **RTCMultiConnection-v1.3 and socket.io** | ---- | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-Demos/RTCMultiConnection-v1.3-and-socket.io.html) |
 
 ----
 
 #### Browser Support
 
-[RTCMultiConnection-v1.3.js](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js) supports following browsers:
+[RTCMultiConnection-v1.3.js](https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js) supports following browsers:
 
 | Browser        | Support           |
 | ------------- |:-------------|
@@ -535,4 +538,4 @@ io.sockets.on('connection', function (socket) {
 
 #### License
 
-[RTCMultiConnection-v1.3.js](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.3.js) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[RTCMultiConnection-v1.3.js](https://www.webrtc-experiment.com/RTCMultiConnection-v1.3.js) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html
index 1de17a10..355944d6 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html
+++ b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html
@@ -190,7 +190,7 @@
             }
 
             .eject {
-                background-image: url('https://webrtc-experiment.appspot.com/images/eject.png') !important;
+                background-image: url('https://www.webrtc-experiment.com/images/eject.png') !important;
                 background-position: center center !important;
                 background-repeat: no-repeat !important;
                 border-color: white;
@@ -208,11 +208,11 @@
             document.createElement('footer');
         </script>
         
-		<script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"> </script>
+		<script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"> </script>
     </head>
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑ WEBRTC EXPERIMENTS</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑ WEBRTC EXPERIMENTS</a>
 
             <h1>
                 <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">RTCMultiConnection-v1.4</a>
@@ -224,7 +224,7 @@ <h1>
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <section>
                 <h2>Open New Session:</h2>
@@ -436,12 +436,12 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                     href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Renegotiation.html b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Renegotiation.html
index 344f072f..3b1c45bb 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Renegotiation.html
+++ b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Renegotiation.html
@@ -122,8 +122,8 @@
                 padding: .4em .8em;
             }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"> </script>
     </head>
     <body>
         <button id="new-session">
@@ -140,7 +140,7 @@
         <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>
+        <a href="https://www.webrtc-experiment.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">
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html
index 2bd422f7..b708189a 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html
+++ b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/Video-Conferencing.html
@@ -95,8 +95,8 @@
 
             footer { text-align: center; }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"> </script>
     </head>
     <body>
         <h1>
@@ -107,7 +107,7 @@ <h1>
                                               Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <section>
             <h2>Open new session:</h2>
@@ -179,12 +179,12 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                     href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html
index 8d5d17dd..92084280 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html
+++ b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/multi-streams-attachment.html
@@ -1,7 +1,7 @@
-<h1>Multi-streams attachment using <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection">RTCMultiConnection</a> / <a href="https://webrtc-experiment.appspot.com/">Home</a></h1>
+<h1>Multi-streams attachment using <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection">RTCMultiConnection</a> / <a href="https://www.webrtc-experiment.com/">Home</a></h1>
 <style>video {width: 48%;vertical-align: top;}</style>
-<script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"> </script>
+<script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"> </script>
 
 <button id="init">Open New Connection</button>
 <br /><br />
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/All-in-One.html b/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/All-in-One.html
index ca304d07..7934d996 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/All-in-One.html
+++ b/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/All-in-One.html
@@ -190,7 +190,7 @@
             }
 
             .eject {
-                background-image: url('https://webrtc-experiment.appspot.com/images/eject.png') !important;
+                background-image: url('https://www.webrtc-experiment.com/images/eject.png') !important;
                 background-position: center center !important;
                 background-repeat: no-repeat !important;
                 border-color: white;
@@ -207,11 +207,11 @@
             document.createElement('article');
             document.createElement('footer');
         </script>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.5.js"> </script>
     </head>
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑ WEBRTC EXPERIMENTS</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑ WEBRTC EXPERIMENTS</a>
 
             <h1>
                 <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">RTCMultiConnection-v1.5</a>
@@ -223,7 +223,7 @@ <h1>
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <section>
                 <h2>Open New Session:</h2>
@@ -432,12 +432,12 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                     href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Renegotiation.html b/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Renegotiation.html
index f84198dd..0653202c 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Renegotiation.html
+++ b/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Renegotiation.html
@@ -122,8 +122,8 @@
                 padding: .4em .8em;
             }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.5.js"> </script>
     </head>
     <body>
         <button id="new-session">
@@ -140,7 +140,7 @@
         <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>
+        <a href="https://www.webrtc-experiment.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">
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Video-Conferencing.html b/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Video-Conferencing.html
index 04c9ee18..89cd3d15 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Video-Conferencing.html
+++ b/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Video-Conferencing.html
@@ -95,8 +95,8 @@
 
             footer { text-align: center; }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.5.js"> </script>
     </head>
     <body>
         <h1>
@@ -107,7 +107,7 @@ <h1>
                                               Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <section>
             <h2>Open new session:</h2>
@@ -179,12 +179,12 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                     href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.5-experimental.md b/RTCMultiConnection/RTCMultiConnection-v1.5-experimental.md
index 81855188..f672d374 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.5-experimental.md
+++ b/RTCMultiConnection/RTCMultiConnection-v1.5-experimental.md
@@ -1,4 +1,4 @@
-##### [RTCMultiConnection-v1.5.js](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5.js) Documentation (Experimental Release)
+##### [RTCMultiConnection-v1.5.js](https://www.webrtc-experiment.com/RTCMultiConnection-v1.5.js) Documentation (Experimental Release)
 
 Supports features like
 
@@ -25,7 +25,7 @@ and much more.
 ##### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.5.js"></script>
 ```
 
 ##### Second Step: Start using it!
@@ -566,19 +566,19 @@ io.sockets.on('connection', function (socket) {
 
 =
 
-##### Demos using [RTCMultiConnection-v1.5](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5.js) library
+##### Demos using [RTCMultiConnection-v1.5](https://www.webrtc-experiment.com/RTCMultiConnection-v1.5.js) library
 
 | Experiment Name        | Demo           | Source Code |
 | ------------- |-------------|-------------|
-| **All-in-One test** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5-Demos/All-in-One.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/All-in-One.html) |
-| **Renegotiation & Mute/UnMute/Stop** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5-Demos/Renegotiation.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Renegotiation.html) |
-| **Video-Conferencing** | [Demo](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5-Demos/Video-Conferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Video-Conferencing.html) |
+| **All-in-One test** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.5-Demos/All-in-One.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/All-in-One.html) |
+| **Renegotiation & Mute/UnMute/Stop** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.5-Demos/Renegotiation.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Renegotiation.html) |
+| **Video-Conferencing** | [Demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.5-Demos/Video-Conferencing.html) | [Source](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.5-Demos/Video-Conferencing.html) |
 
 =
 
 ##### Browser Support
 
-[RTCMultiConnection-v1.5.js](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5.js) supports following browsers:
+[RTCMultiConnection-v1.5.js](https://www.webrtc-experiment.com/RTCMultiConnection-v1.5.js) supports following browsers:
 
 | Browser        | Support           |
 | ------------- |:-------------|
@@ -590,4 +590,4 @@ io.sockets.on('connection', function (socket) {
 
 ##### License
 
-[RTCMultiConnection-v1.5.js](https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.5.js) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[RTCMultiConnection-v1.5.js](https://www.webrtc-experiment.com/RTCMultiConnection-v1.5.js) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/RTCMultiConnection/index.html b/RTCMultiConnection/index.html
index 9aeeabd3..355944d6 100644
--- a/RTCMultiConnection/index.html
+++ b/RTCMultiConnection/index.html
@@ -1,8 +1,7 @@
 <!DOCTYPE html>
 <html id="home" lang="en">
-    
     <head>
-        <title>RTCMultiConnection All-in-One Test ® Muaz Khan</title>
+        <title>RTCMultiConnection-v1.4 All-in-One Test ® Muaz Khan</title>
         <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
         <meta charset="utf-8">
         <meta name="viewport" content="width=device-width, initial-scale=1.0, user-scalable=no">
@@ -12,84 +11,84 @@
         <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();
-			}
+                location.href = location.href + '#' + ((Math.random() * new Date().getTime()).toString(36).toUpperCase().replace( /\./g , '-'));
+                location.reload();
+            }
         </script>
         <style>
             @import url(https://fonts.googleapis.com/css?family=Inconsolata);
-            html {
-                background: #eee;
-            }
+
+            html { background: #eee; }
+
             body {
-                margin: 0;
-                font-family:"Inconsolata", "DejaVu Sans Mono", "Bitstream Vera Sans Mono", monospace;
+                font-family: "Inconsolata", "DejaVu Sans Mono", "Bitstream Vera Sans Mono", monospace;
                 font-size: 1.2em;
                 line-height: 1.2em;
+                margin: 0;
             }
+
             article, footer {
                 display: block;
-                min-width: 360px;
                 max-width: 900px;
+                min-width: 360px;
                 width: 80%;
             }
+
             article {
-                margin: 2.5em auto 0 auto;
+                background: #fff;
                 border: 1px solid;
                 border-color: #ddd #aaa #aaa #ddd;
+                margin: 2.5em auto 0 auto;
                 padding: 2em;
-                background: #fff;
-            }
-            h1 {
-                margin-top: 0;
-            }
-            article p:first-of-type {
-                margin-top: 1.6em;
-            }
-            article p:last-child {
-                margin-bottom: 0;
             }
+
+            h1 { margin-top: 0; }
+
+            article p:first-of-type { margin-top: 1.6em; }
+
+            article p:last-child { margin-bottom: 0; }
+
             footer {
                 margin: 0 auto 2em auto;
                 text-align: center;
             }
+
             footer a {
                 color: #666;
-                text-shadow: 0 1px 1px #fff;
-                text-decoration: none;
                 font-size: inherit;
                 padding: 1em;
+                text-decoration: none;
+                text-shadow: 0 1px 1px #fff;
             }
-            footer a:hover, footer a:focus {
-                color: #111;
-            }
+
+            footer a:hover, footer a:focus { color: #111; }
+
             h1, h2 {
-                font-weight: normal;
-                display: inline;
                 border-bottom: 1px solid black;
-                padding: 0 0 3px 0;
+                display: inline;
+                font-weight: normal;
                 line-height: 36px;
+                padding: 0 0 3px 0;
             }
+
             a {
                 color: #2844FA;
                 text-decoration: none;
             }
-            a:hover, a:focus {
-                color: #1B29A4;
-            }
-            a:active {
-                color: #000;
-            }
+
+            a:hover, a:focus { color: #1B29A4; }
+
+            a:active { color: #000; }
+
             :-moz-any-link:focus {
-                color: #000;
                 border: 0;
+                color: #000;
             }
-            ::selection {
-                background: #ccc;
-            }
-            ::-moz-selection {
-                background: #ccc;
-            }
+
+            ::selection { background: #ccc; }
+
+            ::-moz-selection { background: #ccc; }
+
             button, select {
                 -moz-border-radius: 3px;
                 -moz-transition: none;
@@ -101,100 +100,106 @@
                 border-radius: 3px;
                 color: #fff;
                 display: inline-block;
+                font-family: inherit;
                 font-size: .8em;
                 line-height: 1.3;
                 padding: 5px 12px;
                 text-align: center;
                 text-shadow: 1px 1px 1px #076bd2;
-                font-family: inherit;
-            }
-            button:hover {
-                background: rgb(9, 147, 240);
-            }
-            button:active {
-                background: rgb(10, 118, 190);
             }
+
+            button:hover { background: rgb(9, 147, 240); }
+
+            button:active { background: rgb(10, 118, 190); }
+
             button[disabled] {
                 background: none;
                 border: 1px solid rgb(187, 181, 181);
                 color: gray;
                 text-shadow: none;
             }
+
             strong {
                 color: rgb(204, 14, 14);
                 font-family: inherit;
                 font-weight: normal;
             }
+
             #result div {
-                padding: .8em 1.5em;
                 border: 1px solid Gray;
                 border-bottom: 0;
+                padding: .8em 1.5em;
             }
+
             select {
+                background: transparent;
                 color: black;
-                text-shadow: none;
                 text-align: left;
-                background: transparent;
-            }
-            #remote-media-streams video {
-                width: 10em;
-            }
-            #local-media-stream video {
-                width: 34em;
-            }
-            td {
-                vertical-align: top;
+                text-shadow: none;
             }
+
+            #remote-media-streams video { width: 10em; }
+
+            #local-media-stream video { width: 34em; }
+
+            td { vertical-align: top; }
+
             #chat-output div {
                 border: 1px solid black;
                 border-bottom: 0;
                 padding: .1em .4em;
             }
+
             input {
                 border: 1px solid black;
-                outline: none;
-                margin: .1em .3em;
                 font-family: inherit;
+                margin: .1em .3em;
+                outline: none;
                 padding: .1em .2em;
                 width: 93%;
             }
+
             #chat-output {
+                margin: 0 0 0 .4em;
                 max-height: 12em;
                 overflow: auto;
-                margin: 0 0 0 .4em;
             }
+
             pre {
                 border-left: 2px solid red;
                 margin-left: 2em;
                 padding-left: 1em;
             }
+
             .video-container {
-                display: inline-block;
+                background: white;
                 border: 2px solid black;
-                padding: .1em;
                 border-radius: 0.2em;
+                display: inline-block;
                 margin: 2em .2em;
-                background: white;
+                padding: .1em;
                 vertical-align: top;
             }
+
             .video-container h2 {
                 border: 0;
                 border-top: 1px solid black;
+                display: block;
                 margin: 0;
                 text-align: center;
-                display: block;
             }
+
             .eject {
+                background-image: url('https://www.webrtc-experiment.com/images/eject.png') !important;
                 background-position: center center !important;
                 background-repeat: no-repeat !important;
-                background-image: url('/images/eject.png') !important;
-                width: 30px;
+                border-color: white;
+                border-radius: 0;
                 height: 30px;
+                margin: .2em;
                 position: absolute;
+                width: 30px;
                 z-index: 200;
-                margin: .2em;
-                border-color: white;
-                border-radius: 0;
             }
         </style>
         <!-- for HTML5 el styling -->
@@ -202,38 +207,43 @@
             document.createElement('article');
             document.createElement('footer');
         </script>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js"></script>
+        
+		<script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"> </script>
     </head>
-    
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none; right: 0; top: 0;">↑ WEBRTC EXPERIMENTS</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑ WEBRTC EXPERIMENTS</a>
 
-             <h1>
-            <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">RTCMultiConnection</a>
-            all-in-one test
-        </h1>
+            <h1>
+                <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">RTCMultiConnection-v1.4</a>
+                all-in-one test
+            </h1>
 
             <p> <span>Copyright © 2013</span>  <a href="https://github.com/muaz-khan" target="_blank">Muaz
-    Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
+                                                   Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <section>
-                 <h2>Open New Session:</h2>
+                <h2>Open New Session:</h2>
 
                 <select id="session" title="Session">
-                    <option>audio + video + data</option>
-                    <option>only audio and data</option>
-                    <option>only video and data</option>
-                    <option>audio and video</option>
-                    <option>only audio</option>
-                    <option>only video</option>
-                    <option>only data</option>
-                    <option>only screen</option>
-                    <option>only screen and data</option>
+					<option>audio+video+data+screen</option>
+                    <option>audio+video+data</option>
+					<option>audio+data+screen</option>
+					<option>audio+video+screen</option>
+                    <option>audio+video</option>
+					<option>audio+screen</option>
+					<option>video+screen</option>
+					<option>data+screen</option>
+                    <option>audio+data</option>
+                    <option>video+data</option>
+                    <option>audio</option>
+                    <option>video</option>
+                    <option>data</option>
+                    <option>screen</option>
                 </select>
                 <select id="direction" title="Direction">
                     <option>many-to-many</option>
@@ -243,19 +253,19 @@ <h2>Open New Session:</h2>
                 </select>
                 <button id="open-session" disabled>Open NEW Session</button>
             </section>
-            <table style="width: 100%; border-left: 1px solid black;">
+            <table style="border-left: 1px solid black; width: 100%;">
                 <tr>
                     <td style="width: 80%;">
-                         <h2 style="display: block; text-align: center;">Local Media Stream</h2>
+                        <h2 style="display: block; text-align: center;">Local Media Stream</h2>
 
                         <section id="local-media-stream"></section>
                     </td>
-                    <td style="border-left: 1px solid black; background: white;">
-                         <h2 style="display: block; font-size: 1em; text-align: center;">Share Files</h2>
+                    <td style="background: white; border-left: 1px solid black;">
+                        <h2 style="display: block; font-size: 1em; text-align: center;">Share Files</h2>
 
                         <input type="file" id="file" disabled>
                         <br />
-                         <h2 style="display: block; font-size: 1em; text-align: center;">Text Chat</h2>
+                        <h2 style="display: block; font-size: 1em; text-align: center;">Text Chat</h2>
 
                         <div id="chat-output"></div>
                         <input type="text" id="chat-input" style="font-size: 1.2em;" placeholder="chat message" disabled>
@@ -263,7 +273,7 @@ <h2 style="display: block; font-size: 1em; text-align: center;">Text Chat</h2>
                 </tr>
                 <tr>
                     <td colspan="2">
-                         <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
+                        <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
 
                         <section id="remote-media-streams"></section>
                     </td>
@@ -271,64 +281,85 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
             </table>
             <script>
                 function initConnection(config) {
-                    window.connection = new RTCMultiConnection(config.sessionid, {
-                        firebase: 'rtcweb',
-                        session: config.session,
-                        direction: config.direction
-                    });
-                    connection.onopen = function () {
+                    window.connection = new RTCMultiConnection(config.sessionid);
+
+                    connection.firebase = 'signaling';
+                    connection.session = config.session;
+
+                    connection.maxParticipantsAllowed = config.maxParticipantsAllowed;
+
+                    connection.onopen = function() {
                         document.getElementById('chat-input').disabled = false;
                         document.getElementById('file').disabled = false;
-                        if (document.getElementById('open-session')) document.getElementById('open-session').disabled = true;
+                        if (document.getElementById('open-session'))
+                            document.getElementById('open-session').disabled = true;
+                    };
+                    connection.onmessage = function(e) {
+                        appendDIV(e.data);
                     };
-                    connection.onmessage = appendDIV;
-                    connection.onstream = function (stream) {
-                        var video = getVideo(stream);
-                        if (stream.type === 'local') document.getElementById('local-media-stream').appendChild(video);
+                    connection.onstream = function(e) {
+                        var video = getVideo(e);
+                        if (e.type === 'local') document.getElementById('local-media-stream').appendChild(video);
 
-                        if (stream.type === 'remote') {
-                            if (stream.direction !== RTCDirection.OneWay) {
+                        if (e.type === 'remote') {
+                            if (!e.session.oneway) {
                                 var remoteMediaStreams = document.getElementById('remote-media-streams');
                                 remoteMediaStreams.appendChild(video, remoteMediaStreams.firstChild);
                             } else document.getElementById('local-media-stream').appendChild(video);
                         }
-                        stream.mediaElement.width = innerWidth / 3.4;
+                        e.mediaElement.width = innerWidth / 3.4;
                     };
 
-                    connection.onFileProgress = function (packets) {
+                    connection.onFileProgress = function(packets) {
                         appendDIV(packets.remaining + ' packets remaining.');
                         if (packets.sent) appendDIV(packets.sent + ' packets sent.');
                         if (packets.received) appendDIV(packets.received + ' packets received.');
                     };
 
-                    connection.onFileReceived = function (fileName) {
+                    connection.onFileReceived = function(fileName) {
                         appendDIV(fileName + ' received.');
                     };
 
-                    connection.onFileSent = function (file) {
+                    connection.onFileSent = function(file) {
                         appendDIV(file.name + ' sent.');
                     };
 
-                    connection.onleave = function (userid) {
-                        var video = document.getElementById(userid);
-                        if (video) video.parentNode.removeChild(video);
+                    connection.onstreamended = function(e) {
+                        if (e.mediaElement.parentNode && e.mediaElement.parentNode.parentNode) e.mediaElement.parentNode.parentNode.removeChild(e.mediaElement.parentNode);
                     };
+
+                    connection.connect(config.sessionid);
                 }
 
-                document.getElementById('open-session').onclick = function () {
+                document.getElementById('open-session').onclick = function() {
+                    var direction = document.getElementById('direction').value;
+                    var _session = document.getElementById('session').value;
+                    var splittedSession = _session.split('+');
+
+                    var session = { };
+                    for (var i = 0; i < splittedSession.length; i++) {
+                        session[splittedSession[i]] = true;
+                    }
+
+                    var maxParticipantsAllowed = 10;
+
+                    if (direction == 'one-to-one') maxParticipantsAllowed = 1;
+                    if (direction == 'one-to-many') session.broadcast = true;
+                    if (direction == 'one-way') session.oneway = true;
+
                     initConnection({
                         sessionid: hash,
-                        session: document.getElementById('session').value,
-                        direction: document.getElementById('direction').value
+                        session: session,
+                        maxParticipantsAllowed: maxParticipantsAllowed
                     });
-                    connection.open();
+                    connection.open(hash);
 
                     this.disabled = true;
                     this.parentNode.innerHTML = '<h2><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
                     window.isRoomInitiator = true;
                 };
 
-                document.getElementById('file').onchange = function () {
+                document.getElementById('file').onchange = function() {
                     var file = this.files[0];
                     connection.send(file);
                 };
@@ -344,8 +375,9 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
 
                     chatInput.focus();
                 }
+
                 var chatInput = document.getElementById('chat-input');
-                chatInput.onkeypress = function (e) {
+                chatInput.onkeypress = function(e) {
                     if (e.keyCode !== 13 || !this.value) return;
                     appendDIV(this.value);
                     connection.send(this.value);
@@ -354,18 +386,18 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
                     this.focus();
                 };
 
-                function getVideo(stream) {
+                function getVideo(d) {
                     var div = document.createElement('div');
                     div.className = 'video-container';
-                    div.id = stream.userid || 'self';
+                    div.id = d.userid || 'self';
 
-                    if (stream.type === 'remote') {
+                    if (d.type === 'remote') {
                         if (window.isRoomInitiator) {
                             var eject = document.createElement('button');
                             eject.className = 'eject';
                             eject.title = 'Eject this User';
 
-                            eject.onclick = function () {
+                            eject.onclick = function() {
                                 // eject a specific user
                                 connection.eject(this.parentNode.id);
                                 this.parentNode.style.display = 'none';
@@ -373,90 +405,43 @@ <h2 style="display: block; text-align: center;">Remote Media Streams</h2>
                             div.appendChild(eject);
                         }
                     }
-                    stream.mediaElement.controls = true;
-                    stream.mediaElement.autoplay = true;
-                    div.appendChild(stream.mediaElement);
-                    return div
+                    div.appendChild(d.mediaElement);
+                    return div;
                 }
 
                 console.debug('about to check presence of a room');
-                    new window.Firebase('https://rtcweb.firebaseIO.com/' + hash).once('value', function (data) {
-                        var isRoomPresent = data.val() != null;
-                        if (isRoomPresent) {
-                            initConnection({
-                                sessionid: hash,
-                                direction: 'many-to-many',
-                                session: 'audio-video'
-                            });
-                            document.getElementById('open-session').disabled = true;
-                            window.isRoomInitiator = false;
-                        } else document.getElementById('open-session').disabled = false;
-
-                        console.debug('room is present? ', isRoomPresent);
-                    });
+                new window.Firebase('https://signaling.firebaseIO.com/' + hash).once('value', function(data) {
+                    var isRoomPresent = data.val() != null;
+                    if (isRoomPresent) {
+                        initConnection({
+                            sessionid: hash
+                        });
+                        document.getElementById('open-session').disabled = true;
+                        window.isRoomInitiator = false;
+                    } else document.getElementById('open-session').disabled = false;
+
+                    console.debug('room is present? ', isRoomPresent);
+                });
             </script>
             <br />
             <br />
-             <h2>Getting started with <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection"
-            target="_blank">RTCMultiConnection</a></h2>
- <pre>
-&lt;script src="<a href="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js" target="_blank">https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.2.js</a>"&gt;&lt;/script&gt;
-&lt;script&gt;
-    var connection = new <strong>RTCMultiConnection</strong>();
-    connection.<strong>open</strong>('session-id');    // open a new session
-    connection.<strong>connect</strong>('session-id'); // connect a pre-created session
-
-    connection.eject('user-id');                // eject a specific user
-    connection.leave();                         // leave the session
-    connection.onleave = function(userid) { }   // if someone leaves
-&lt;/script&gt;
-</pre>
-
-            <br />
-            <br />
-             <h2><a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">RTCMultiConnection</a>
-            features:</h2>
-
-            <ol>
-                <li>Users ejection, rejection and presence detection support</li>
-                <li>Multi-sessions establishment as well as manual-sessions establishment</li>
-                <li>Sessions will be active all the time; even if initiator leaves!</li>
-                <li>Data i.e. Text/File sharing (of any size)</li>
-                <li>Screen sharing, audio-video conferencing; file sharing and one-way broadcasting</li>
-            </ol>
-            <br/>
-            <br/>
-             <h2>What is <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">RTCMultiConnection</a>?</h2>
-
-            <br/>
-            <br/>RTCMultiConnection.js a javascript library supports features like audio/video conferencing; (one-way and one-to-many) broadcasting; screen sharing; data/text/file sharing (of any size); multi-and-manual sessions establishment; users ejection, rejection
-            and presence detection; and more. It automatically keeps session "active" all the time; even if initiator leaves.
-            <br />
-            <br />
-             <h2>
-            <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">Source code
-        and Documentation on Github!</a>
-        </h2>
-
-            <br />
-            <br />
-            <section style="border: 1px solid rgb(189, 189, 189); margin: 1em 3em; border-radius: .2em;">
-                 <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 189, 189);">Feedback</h2>
+             
+            <section style="border: 1px solid rgb(189, 189, 189); border-radius: .2em; margin: 1em 3em;">
+                <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .2em .4em;">Feedback</h2>
 
                 <div>
-                    <textarea id="message" style="height: 8em; margin: .2em; width: 98%; border: 1px solid rgb(189, 189, 189); outline: none; resize: vertical;" placeholder="Have any message? Suggestions or something went wrong?"></textarea>
+                    <textarea id="message" style="border: 1px solid rgb(189, 189, 189); height: 8em; margin: .2em; outline: none; resize: vertical; width: 98%;" placeholder="Have any message? Suggestions or something went wrong?"></textarea>
                 </div>
                 <button id="send-message" style="font-size: 1em;">Send Message</button>
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
-                href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
+                    href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
-
-</html>
\ No newline at end of file
+</html>
diff --git a/RTCPeerConnection/README.md b/RTCPeerConnection/README.md
index e205be51..f70d53bd 100644
--- a/RTCPeerConnection/README.md
+++ b/RTCPeerConnection/README.md
@@ -1,115 +1,166 @@
 ##### RTCPeerConnection Documentation
 
-**RTCPeerConnection.js** is a JavaScript wrapper library for **RTCWeb PeerConnection APIs**.
+**RTCPeerConnection.js** is a js-wrapper library for **RTCWeb APIs**. 
 
-It not only simplifies coding but also handles complex cross browser implementations.
-
-=
-
-##### Getting started with RTCPeerConnection library
-
-First of all; library to be linked!
+It not only simplifies coding but also handles complex cross browser implementations. 
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.6.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
 ```
 
 **RTCPeerConnection** object's structure looks like this:
 
 ```javascript
-var peer = RTCPeerConnection({
-    attachStream: localMediaStream,
-    offerSDP: RTCSessionDescription,
+var peer = RTCPeerConnection(configuration);
+```
+
+Actually you're calling a method; and passing some data over it; and that's it!
+
+`configuration` is an object contains a few properties and methods:
+
+```javascript
+var configuration = {
+    attachStream: MediaStream,
+    attachStreams: [MediaStream_1, MediaStream_2, MediaStream_3],
+	
+    offerSDP: offerSDP_sent_from_offerer,
 
-    onRemmoteSteam: function (stream) {},
+    onICE: function (candidate) {},
+    onRemoteStream: function (stream) {},
+    onRemoteStreamEnded: function (stream) {},
 
     onOfferSDP: function (offerSDP) {},
     onAnswerSDP: function (answerSDP) {},
 
     onChannelMessage: function (event) {},
     onChannelOpened: function (_RTCDataChannel) {}
-});
+};
 ```
 
 =
 
-##### Exploring RTCPeerConnection object
+#### attachStream
 
-You're passing a configuration object that contains a few methods and objects.
+The `MediaStream` you want to attach. To understand it better; `RTCPeerConnection` will use it like this:
 
-=
+```javascript
+var configuration = {
+    attachStream: MediaStream
+};
+```
 
-##### `attachStream`
+=
 
-This object must contain `LocalMediaStream` object that will be attached to share your audio/video stream. You can skip it if you want to implement one-way streaming.
+#### attachStreams
 
-Behind the scene, **RTCPeerConnection** is calling `addStream` method:
+You can attach multiple streams too; see [this demo](https://www.webrtc-experiment.com/demos/screen-and-video-from-single-peer.html).
 
 ```javascript
-peerConnection.addStream(attachStream);
+var configuration = {
+    attachStreams: [screenStream, audioStream, videoStream]
+};
 ```
 
 =
 
-##### `offerSDP`
+#### offerSDP
 
-In one-to-one media session; this object will be used by **answerer** as soon as he will get **offer sdp** sent from **offerer**. Actually you're adding **offer sdp** here:
+This object is only useful for `answerer`. As soon as you'll receive `offer sdp` sent by `offerer`; use that `offer sdp` like this:
 
 ```javascript
-offerSDP = {
-   type: 'offer',
-   sdp: '------'
+configuration.offerSDP = offerSDP_sent_by_offerer;
+```
+
+Where `offerSDP_sent_by_offerer` MUST be an `object` and MUST look like this:
+
+```javascript
+{
+    type: 'offer',
+    sdp: '.........offerer sdp.........'
 }
 ```
 
 =
 
-##### `onRemmoteSteam`
+#### onICE
+
+`RTCPeerConnection` will call this method on each new ICE candidate. `RTCPeerConnection` will pass `RTCIceCandidate` object over this method:
+
+```javascript
+onICE: function (_RTCIceCandidate) {
+    // _RTCIceCandidate.candidate
+    // _RTCIceCandidate.sdpMLineIndex
+    // etc.
+};
+```
+
+You can use SIP or any other signaling method to send these ICE candidates on the other end.
+
+=
+
+#### onRemoteStream
 
-This method is invoked as soon as **remote media stream** is added in peer connection object.
+`RTCPeerConnection` will call this method as soon as `peer.onaddstream` event will fire. `RTCPeerConnection` will pass `RemoteStream` object over this method:
 
 ```javascript
 onRemmoteSteam: function (remoteMediaStream) {
-    // if(firefox) remoteVideo.mozSrcObject = remoteMediaStream;
-    // if(firefox) remoteVideo.src = window.URL.createObjectURL(remoteMediaStream);
-	
-    // if(chrome) remoteVideo.src = window.webkitURL.createObjectURL(remoteMediaStream);
+    // if(moz) remoteVideo.mozSrcObject = remoteMediaStream;
+    // if(!moz) remoteVideo.src = window.webkitURL.createObjectURL(remoteMediaStream);
 }
 ```
 
 =
 
-##### `onOfferSDP`
+#### onRemoteStreamEnded
 
-Use this method on the **offerer**'s side to get access to the **offer-sdp** generated using `peerConnection.createOffer` method:
+It is fired when remote stream stops flowing.
+
+```javascript
+onRemoteStreamEnded: function (remoteMediaStream) {
+    // var video = document.getElementById(remoteMediaStream.id);
+    // if(video) video.parentNode.removeChild(video);
+}
+```
+
+=
+
+#### onOfferSDP
+
+RTCPeerConnection will call this method after successfully creating offer SDP. An object of type `RTCSessionDescription` will be passed over this method:
 
 ```javascript
 onOfferSDP: function (offerSDP) {
     // offerSDP.type === 'offer'
     // offerSDP.sdp
+    // to POST using XHR: JSON.stringify(offerSDP)
 }
 ```
 
+Pass offer sdp on the answerer's side using your own preferred signaling method.
+
 =
 
-##### `onAnswerSDP`
+#### onAnswerSDP
 
-As soon as offerer sends you **offer-sdp** using his preferred signaling method; you should initiate peer connection for **answerer**. That peer connection object will create **answer-sdp** accordingly. To get access to that **answer-sdp**, use `onAnswerSDP` method:
+RTCPeerConnection will call this method after successfully creating answer SDP. An object of type `RTCSessionDescription` will be passed over this method:
 
 ```javascript
 onAnswerSDP: function (answerSDP) {
     // answerSDP.type === 'answer'
     // answerSDP.sdp
+    // to POST using XHR: JSON.stringify(answerSDP)
 }
 ```
 
-**RTCPeerConnection** library also allows you establish data connection between peers.
+Pass `answer sdp` on the `offerer's` side using your own preferred signaling method.
 
 =
 
-##### `onChannelMessage`
+#### onChannelMessage
 
-Use this method to get access to all text messages or data transferred over WebRTC data channels.
+Pass this method only if you want to create/open (cross browser RTC) data channels to share data/text or files.
+
+`RTCPeerConnection` object will call this method as soon as new data-message will be received over SCTP or RTP data ports.
 
 ```javascript
 onChannelMessage: function (event) {
@@ -120,9 +171,9 @@ onChannelMessage: function (event) {
 
 =
 
-##### `onChannelOpened`
+#### onChannelOpened
 
-Use this method if you want to be alerted when data ports get open.
+RTCPeerConnection object will call this method as soon as SCTP/RTP data ports will successfully get open.
 
 ```javascript
 onChannelOpened: function (channel) {
@@ -132,44 +183,60 @@ onChannelOpened: function (channel) {
 
 =
 
-##### Public instance methods
-
-**RTCPeerConnection** library has a few instance methods to be called later at appropriate time.
+#### Public instance methods
 
-Here is how to get access to those instance methods:
+RTCPeerConnection object has some public methods/objects that can be used/called later.
 
 ```javascript
 var peer = RTCPeerConnection(configuration);
+```
+
+You can use `peer` object to call/use those instance methods.
 
-// using "peer" object; you can call those instance methods
-peer.addAnswerSDP(...);
+=
+
+#### Add ICE candidates
+
+As soon as you'll receive ICE candidate sent by other peer; add it like this:
+
+```javascript
+peer.addICE({
+    sdpMLineIndex: websocketMessage.sdpMLineIndex,
+    candidate: websocketMessage.candidate
+});
 ```
 
 =
 
-##### `addAnswerSDP`
+#### Add Answer SDP
 
-Use `addAnswerSDP` instance method to add **answer-sdp** sent by answerer. This method should only be used in offerer's side.
+Whenever offerer will receive `answer sdp` sent by answerer; add that `answer sdp` like this:
 
 ```javascript
 peer.addAnswerSDP(answerSDP);
+```
 
-// answer-sdp looks like this:
+Where `answerSDP` must look like this:
+
+```javascript
 {
     type: 'answer',
-    sdp: '-----'
+    sdp: ''
 }
 ```
 
 =
 
-##### How to send data over WebRTC Data Channels?
+#### Send data/text/file over RTCDataChannel
+
+On **Firefox**, you can send files directly; because Firefox supports `binaryType` for `RTCDataChannel`.
+
+On Chrome, `binaryType` will be supported soon.
 
-`peer` object has another instance method named `sendData` that allows you send only text-messages on chrome and blobs, buffers, and objects on firefox.
+You can send text message like this:
 
 ```javascript
 peer.sendData('hi there, I am a text message');
-peer.channel.send('hi there, I am a text message');
 
 /* to send an object */
 var object = {
@@ -180,50 +247,75 @@ var object = {
     _function: function() {}
 };
 var stringified = JSON.stringify(object);
-
 peer.sendData(stringified);
-peer.channel.send(stringified);
+```
 
-// because firefox accepts blobs too, send file directly
-peer.sendData(file);
+You can send same text message like this too:
+
+```javascript
+peer.channel.send(hi there, I am a text message');
 ```
 
-=
+Where `channel` is `RTCDataChannel` object.
+
+On Firefox, you can send file directly like this:
 
-##### `onChannelClosed`
+```javascript
+peer.sendData(file);
+// or otherwise: 
+// peer.channel.send(file);
+```
 
-Use this method to be alerted when data ports get close.
+You can access `[webkit|moz]RTCPeerConnection` object too to add additional events or do extra stuff:
 
 ```javascript
-onChannelClosed: function (event) { }
+var rtcPeerConnection = peer.peer;
+// rtcPeerConnection.getLocalStreams()[0].stop()
+// rtcPeerConnection.getAudioTracks()
+// rtcPeerConnection.getVideoTracks()
 ```
 
 =
 
-##### `onChannelError`
+#### Catching RTCDataChannel Errors
 
-Use this method to catch errors expected to be occur in data channels.
+To get alerted whenever a data port closes or throws an error message:
 
 ```javascript
-onChannelError: function (event) { }
+var configuration = {
+    onChannelClosed: function (event) {},
+    onChannelError: function (event) {}
+};
 ```
 
 =
 
-##### `getUserMedia`
+#### getUserMedia
 
-Use cross-browser `getUserMedia` function to get access to camera/microphone.
+This [js-wrapper for RTCWeb API](https://bit.ly/RTCPeerConnection-v1-4) file also contains a **cross-browser** `getUserMedia` function.
 
 ```javascript
-getUserMedia({
+getUserMedia(mediaConfiguration);
+```
+
+Where `mediaConfiguration` object looks like this:
+
+```javascript
+var mediaConfiguration = {
     video: HTMLAudioElement || HTMLVideoElement,
     constraints: {},
     onsuccess: function (localMediaStream) {},
     onerror: function (event) {}
-});
+};
 ```
 
-Use **constraints** to pass custom constraints like `chromeMediaSource` etc.
+Only `onsuccess` is mandatory. All other three are optional.
+
+You can pass `video` element to play it as soon as `navigator.getUserMedia` API will give access to `LocalMediaStream`.
+
+`constraints` object is very useful when applying custom constraints. 
+
+For example to capture screen in Google Chrome Canary:
 
 ```javascript
 var screen_constraints = {
@@ -233,28 +325,81 @@ var screen_constraints = {
     optional: []
 };
 
-getUserMedia({
+var mediaConfiguration = {
     constraints: {
-        audio: false,
+        audio: false, /* MUST be false! */
         video: screen_constraints
     }
-});
+};
+```
+
+=
+
+#### STUN/TURN servers and customization
+
+RTCPeerConnection object uses `STUN` by default.
+
+To understand it better; see how it is using `STUN` server:
+
+```javascript
+var STUN = {
+    iceServers: [{
+        url: !moz ? 'stun:stun.l.google.com:19302' : 'stun:23.21.150.121'
+    }]
+};
 ```
 
+On chrome; it is using: `stun:stun.l.google.com:19302`
+
+And on Firefox; it is using: `stun:23.21.150.121`
+
+
+`RTCPeerConnection` object will use this `TURN` server:
+
+```javascript
+var TURN = {
+    iceServers: [{
+        url: 'turn:webrtc%40live.com@numb.viagenie.ca',
+        credential: 'muazkh'
+    }]
+};
+```
+
+=
+
+#### You MUST keep in mind that...
+
+1. `TURN` is not yet implemented in **Firefox**.
+2. `RTCDataChannel` is not yet interoperable
+3. Old Firefox sometimes fails on `non-DNS STUN` servers
+
+=
+
+#### How to use RTCPeerConnection?
+
+[Here is a simple, step-by-step guide](https://www.webrtc-experiment.com/docs/how-to-use-rtcpeerconnection-js-v1.1.html) that not only explains how to exchange SDP/ICE but also explains "How to write a realtime one-to-one WebRTC video chatting app using socket.io or WebSockets".
+
+In each WebRTC session; there are two circumstances:
+
+1. One party creates offer
+2. One party creates answer
+
+You just need to exchange offer and answer between them using your preferable signaling method like XHR/socket.io/WebSockets/etc.
+
 =
 
 ##### History
 
-1. [RTCPeerConnection-v1.6.js](https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.6.js)
-2. [RTCPeerConnection-v1.5.js](https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js)
-3. [RTCPeerConnection-v1.4.js](https://webrtc-experiment.appspot.com/lib/RTCPeerConnection-v1.4.js)
-4. [RTCPeerConnection-v1.3.js](https://webrtc-experiment.appspot.com/lib/RTCPeerConnection-v1.3.js)
-5. [RTCPeerConnection-v1.2.js](https://webrtc-experiment.appspot.com/lib/RTCPeerConnection-v1.2.js)
-6. [RTCPeerConnection-v1.1.js](https://webrtc-experiment.appspot.com/lib/RTCPeerConnection-v1.1.js)
-7. [RTCPeerConnection.js](https://webrtc-experiment.appspot.com/RTCPeerConnection.js)
+1. [RTCPeerConnection-v1.6.js](https://www.webrtc-experiment.com/RTCPeerConnection-v1.6.js)
+2. [RTCPeerConnection-v1.5.js](https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js)
+3. [RTCPeerConnection-v1.4.js](https://www.webrtc-experiment.com/lib/RTCPeerConnection-v1.4.js)
+4. [RTCPeerConnection-v1.3.js](https://www.webrtc-experiment.com/lib/RTCPeerConnection-v1.3.js)
+5. [RTCPeerConnection-v1.2.js](https://www.webrtc-experiment.com/lib/RTCPeerConnection-v1.2.js)
+6. [RTCPeerConnection-v1.1.js](https://www.webrtc-experiment.com/lib/RTCPeerConnection-v1.1.js)
+7. [RTCPeerConnection.js](https://www.webrtc-experiment.com/RTCPeerConnection.js)
 
 =
 
 ##### License
 
-**RTCPeerConnection.js** is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+**RTCPeerConnection.js** is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/RTCPeerConnection/RTCPeerConnection-v1.5.js b/RTCPeerConnection/RTCPeerConnection-v1.5.js
index c4ca5796..41f0ba82 100644
--- a/RTCPeerConnection/RTCPeerConnection-v1.5.js
+++ b/RTCPeerConnection/RTCPeerConnection-v1.5.js
@@ -1,11 +1,10 @@
-/*
-     2013, @muazkh - github.com/muaz-khan
-     MIT License - https://webrtc-experiment.appspot.com/licence/
-     Documentation - https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCPeerConnection
-*/
-
-window.moz = !! navigator.mozGetUserMedia;
-var RTCPeerConnection = function (options) {
+// 2013, @muazkh - github.com/muaz-khan
+// MIT License - https://webrtc-experiment.appspot.com/licence/
+// Documentation - https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCPeerConnection
+
+window.moz = !!navigator.mozGetUserMedia;
+
+function RTCPeerConnection(options) {
     var w = window,
         PeerConnection = w.mozRTCPeerConnection || w.webkitRTCPeerConnection,
         SessionDescription = w.mozRTCSessionDescription || w.RTCSessionDescription,
@@ -25,7 +24,7 @@ var RTCPeerConnection = function (options) {
     };
 
     if (!moz && !options.iceServers) {
-        if (parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2]) >= 28)
+        if (parseInt(navigator.userAgent.match( /Chrom(e|ium)\/([0-9]+)\./ )[2]) >= 28)
             TURN = {
                 url: 'turn:turn.bistri.com:80',
                 credential: 'homeo',
@@ -50,27 +49,41 @@ var RTCPeerConnection = function (options) {
             }];
     }
 
-    var peerConnection = new PeerConnection(iceServers, optional);
+    var peer = new PeerConnection(iceServers, optional);
 
     openOffererChannel();
 
-    peerConnection.onicecandidate = onicecandidate;
-    if (options.attachStream) peerConnection.addStream(options.attachStream);
-    peerConnection.onaddstream = onaddstream;
+    peer.onicecandidate = function(event) {
+        if (event.candidate)
+            options.onICE(event.candidate);
+    };
+
+    // attachStream = MediaStream;
+    if (options.attachStream) peer.addStream(options.attachStream);
 
-    function onicecandidate(event) {
-        if (!event.candidate || !peerConnection) return;
-        if (options.onICE) options.onICE(event.candidate);
+    // attachStreams[0] = audio-stream;
+    // attachStreams[1] = video-stream;
+    // attachStreams[2] = screen-capturing-stream;
+    if (options.attachStreams && options.attachStream.length) {
+        var streams = options.attachStreams;
+        for (var i = 0; i < streams.length; i++) {
+            peer.addStream(streams[i]);
+        }
     }
 
-    var remoteStreamEventFired = false;
+    peer.onaddstream = function(event) {
+        var remoteMediaStream = event.stream;
 
-    function onaddstream(event) {
-        info('------------onaddstream');
-        if (remoteStreamEventFired || !event || !options.onRemoteStream) return;
-        remoteStreamEventFired = true;
-        options.onRemoteStream(event.stream);
-    }
+        // onRemoteStreamEnded(MediaStream)
+        remoteMediaStream.onended = function() {
+            if (options.onRemoteStreamEnded) options.onRemoteStreamEnded(remoteMediaStream);
+        };
+
+        // onRemoteStream(MediaStream)
+        if (options.onRemoteStream) options.onRemoteStream(remoteMediaStream);
+
+        console.debug('on:add:stream', remoteMediaStream);
+    };
 
     var constraints = options.constraints || {
         optional: [],
@@ -80,182 +93,175 @@ var RTCPeerConnection = function (options) {
         }
     };
 
-    var chars = 'ABCDEFGHIJKLMNOPQRSTUVWXYZ'.split(''),
-        extractedChars = '';
-
-    function getChars() {
-        extractedChars += chars[parseInt(Math.random() * 40)] || '';
-        if (extractedChars.length < 40) getChars();
-
-        return extractedChars;
-    }
-
-    function getInteropSDP(sdp) {
-        // for audio-only streaming: multiple-crypto lines are not allowed
-        if (options.onAnswerSDP)
-            sdp = sdp.replace(/(a=crypto:0 AES_CM_128_HMAC_SHA1_32)(.*?)(\r\n)/g, '');
-
-
-        var inline = getChars() + '\r\n' + (extractedChars = '');
-        sdp = sdp.indexOf('a=crypto') == -1 ? sdp.replace(/c=IN/g,
-            'a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:' + inline +
-            'c=IN') : sdp;
-
-        sdp = setBandwidth(sdp);
-
-        if (options.offerSDP) {
-            info('\n--------offer sdp provided by offerer\n');
-            info(options.offerSDP.sdp);
-        }
-
-        info(options.onOfferSDP ? '\n--------offer\n' : '\n--------answer\n');
-        info('sdp: ' + sdp);
-
-        return sdp;
-    }
+    // onOfferSDP(RTCSessionDescription)
 
     function createOffer() {
         if (!options.onOfferSDP) return;
 
-        peerConnection.createOffer(function (sessionDescription) {
-            sessionDescription.sdp = getInteropSDP(sessionDescription.sdp);
-            peerConnection.setLocalDescription(sessionDescription);
+        peer.createOffer(function(sessionDescription) {
+            sessionDescription.sdp = serializeSdp(sessionDescription.sdp);
+            peer.setLocalDescription(sessionDescription);
             options.onOfferSDP(sessionDescription);
         }, null, constraints);
     }
 
+    // onAnswerSDP(RTCSessionDescription)
+
     function createAnswer() {
         if (!options.onAnswerSDP) return;
 
-        options.offerSDP = new SessionDescription(options.offerSDP);
-        peerConnection.setRemoteDescription(options.offerSDP);
-
-        peerConnection.createAnswer(function (sessionDescription) {
-            sessionDescription.sdp = getInteropSDP(sessionDescription.sdp);
-            peerConnection.setLocalDescription(sessionDescription);
+        peer.setRemoteDescription(new SessionDescription(options.offerSDP));
+        peer.createAnswer(function(sessionDescription) {
+            sessionDescription.sdp = serializeSdp(sessionDescription.sdp);
+            peer.setLocalDescription(sessionDescription);
             options.onAnswerSDP(sessionDescription);
         }, null, constraints);
     }
 
-    function setBandwidth(sdp) {
-        // Firefox has no support of "b=AS"
-        if (moz) return sdp;
+    // if Mozilla Firefox & DataChannel; offer/answer will be created later
+    if ((options.onChannelMessage && !moz) || !options.onChannelMessage) {
+        createOffer();
+        createAnswer();
+    }
 
+
+    // DataChannel Bandwidth
+
+    function setBandwidth(sdp) {
         // remove existing bandwidth lines
-        sdp = sdp.replace(/b=AS([^\r\n]+\r\n)/g, '');
+        sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , '');
+        sdp = sdp.replace( /a=mid:data\r\n/g , 'a=mid:data\r\nb=AS:1638400\r\n');
+
+        return sdp;
+    }
+
+    // old: FF<>Chrome interoperability management
+
+    function getInteropSDP(sdp) {
+        var chars = 'ABCDEFGHIJKLMNOPQRSTUVWXYZ'.split(''),
+            extractedChars = '';
+
+        function getChars() {
+            extractedChars += chars[parseInt(Math.random() * 40)] || '';
+            if (extractedChars.length < 40)
+                getChars();
+
+            return extractedChars;
+        }
+
+        // for audio-only streaming: multiple-crypto lines are not allowed
+        if (options.onAnswerSDP)
+            sdp = sdp.replace( /(a=crypto:0 AES_CM_128_HMAC_SHA1_32)(.*?)(\r\n)/g , '');
 
-        sdp = sdp.replace(/a=mid:audio\r\n/g, 'a=mid:audio\r\nb=AS:50\r\n');
-        sdp = sdp.replace(/a=mid:video\r\n/g, 'a=mid:video\r\nb=AS:256\r\n');
-        sdp = sdp.replace(/a=mid:data\r\n/g, 'a=mid:data\r\nb=AS:1638400\r\n');
+        var inline = getChars() + '\r\n' + (extractedChars = '');
+        sdp = sdp.indexOf('a=crypto') == -1 ? sdp.replace( /c=IN/g ,
+            'a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:' + inline +
+                'c=IN') : sdp;
 
         return sdp;
     }
 
-    if ((options.onChannelMessage && !moz) || !options.onChannelMessage) {
-        createOffer();
-        createAnswer();
+    function serializeSdp(sdp) {
+        if (!moz) sdp = setBandwidth(sdp);
+        sdp = getInteropSDP(sdp);
+        console.debug(sdp);
+        return sdp;
     }
 
+    // DataChannel management
     var channel;
 
     function openOffererChannel() {
-        if (!options.onChannelMessage || (moz && !options.onOfferSDP)) return;
+        if (!options.onChannelMessage || (moz && !options.onOfferSDP))
+            return;
 
         _openOffererChannel();
 
-        if (moz && !options.attachStream) {
+        if (moz) {
             navigator.mozGetUserMedia({
-                audio: true,
-                fake: true
-            }, function (stream) {
-                peerConnection.addStream(stream);
-                createOffer();
-            }, useless);
+                    audio: true,
+                    fake: true
+                }, function(stream) {
+                    peer.addStream(stream);
+                    createOffer();
+                }, useless);
         }
     }
 
     function _openOffererChannel() {
-        channel = peerConnection.createDataChannel(
-            options.channel || 'RTCDataChannel',
-            moz ? {} : {
-                reliable: false
-            });
+        channel = peer.createDataChannel(options.channel || 'RTCDataChannel', moz ? { } : {
+            reliable: false
+        });
 
-        if (moz) channel.binaryType = 'blob';
+        if (moz)
+            channel.binaryType = 'blob';
         setChannelEvents();
     }
 
     function setChannelEvents() {
-        channel.onmessage = function (event) {
+        channel.onmessage = function(event) {
             if (options.onChannelMessage) options.onChannelMessage(event);
         };
 
-        channel.onopen = function () {
+        channel.onopen = function() {
             if (options.onChannelOpened) options.onChannelOpened(channel);
         };
-        channel.onclose = function (event) {
+        channel.onclose = function(event) {
             if (options.onChannelClosed) options.onChannelClosed(event);
 
             console.warn('WebRTC DataChannel closed', event);
         };
-        channel.onerror = function (event) {
-            console.error(event);
+        channel.onerror = function(event) {
             if (options.onChannelError) options.onChannelError(event);
         };
     }
 
-    if (options.onAnswerSDP && moz) openAnswererChannel();
+    if (options.onAnswerSDP && moz && options.onChannelMessage)
+        openAnswererChannel();
 
     function openAnswererChannel() {
-        peerConnection.ondatachannel = function (_channel) {
-            channel = _channel;
+        peer.ondatachannel = function(event) {
+            channel = event.channel;
             channel.binaryType = 'blob';
             setChannelEvents();
         };
 
-        if (moz && !options.attachStream) {
+        if (moz) {
             navigator.mozGetUserMedia({
-                audio: true,
-                fake: true
-            }, function (stream) {
-                peerConnection.addStream(stream);
-                createAnswer();
-            }, useless);
+                    audio: true,
+                    fake: true
+                }, function(stream) {
+                    peer.addStream(stream);
+                    createAnswer();
+                }, useless);
         }
     }
 
-    function useless() {}
-
-    function info(information) {
-        console.log(information);
+    function useless() {
     }
 
     return {
-        addAnswerSDP: function (sdp) {
-            info('--------adding answer sdp:');
-            info(sdp.sdp);
-
-            sdp = new SessionDescription(sdp);
-            peerConnection.setRemoteDescription(sdp);
+        addAnswerSDP: function(sdp) {
+            peer.setRemoteDescription(new SessionDescription(sdp));
         },
-        addICE: function (candidate) {
-            info(candidate.candidate);
-            peerConnection.addIceCandidate(new IceCandidate({
+        addICE: function(candidate) {
+            peer.addIceCandidate(new IceCandidate({
                 sdpMLineIndex: candidate.sdpMLineIndex,
                 candidate: candidate.candidate
             }));
         },
 
-        peer: peerConnection,
+        peer: peer,
         channel: channel,
-        sendData: function (message) {
+        sendData: function(message) {
             channel && channel.send(message);
         }
     };
-};
+}
 
+// getUserMedia
 var video_constraints = {
-    mandatory: {},
+    mandatory: { },
     optional: []
 };
 
@@ -264,11 +270,11 @@ function getUserMedia(options) {
         media;
     n.getMedia = n.webkitGetUserMedia || n.mozGetUserMedia;
     n.getMedia(options.constraints || {
-        audio: true,
-        video: video_constraints
-    }, streaming, options.onerror || function (e) {
-        console.error(e);
-    });
+            audio: true,
+            video: video_constraints
+        }, streaming, options.onerror || function(e) {
+            console.error(e);
+        });
 
     function streaming(stream) {
         var video = options.video;
diff --git a/RTCall/README.md b/RTCall/README.md
index 2de5624a..d1f6e3e4 100644
--- a/RTCall/README.md
+++ b/RTCall/README.md
@@ -1,4 +1,4 @@
-#### [RTCall.js](https://webrtc-experiment.appspot.com/RTCall/) — A library for Browser-to-Browser audio-only calling
+#### [RTCall.js](https://www.webrtc-experiment.com/RTCall/) — A library for Browser-to-Browser audio-only calling
 
 Why RTCall.js?
 
@@ -7,14 +7,16 @@ Why RTCall.js?
 
 The entire process is browser to browser.
 
-----
+=
 
 #### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RTCall.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCall.js"></script>
 ```
 
+=
+
 #### Second Step: Start using it!
 
 Remember: All lines are optional!
@@ -23,6 +25,8 @@ Remember: All lines are optional!
 call = new RTCall();
 ```
 
+=
+
 #### `onincomingcall`
 
 `onincomingcall` fires each time if someone calls you:
@@ -33,6 +37,8 @@ call.onincomingcall = function(caller) {
 };
 ```
 
+=
+
 #### `oncustomer`
 
 `oncustomer` is fired only for `admin`:
@@ -43,6 +49,8 @@ call.oncustomer = function(customer) {
 };
 ```
 
+=
+
 #### `onstream`
 
 `onstream` returns remote media stream:
@@ -58,6 +66,8 @@ call.onstream = function(e) {
 };
 ```
 
+=
+
 #### `init`
 
 initializing `RTCall` object:
@@ -66,6 +76,8 @@ initializing `RTCall` object:
 call.init();
 ```
 
+=
+
 #### call
 
 Admin can call customers using their caller-ids. Remeber, customers caller-ids are always passed over `oncustomer` method:
@@ -82,6 +94,8 @@ Customers can call the admin too; using admin's `caller-id`:
 call.call('admin-caller-id');
 ```
 
+=
+
 #### `receive`
 
 `receiver-id` is always passed over `onincomingcall`:
@@ -92,6 +106,8 @@ call.onincomingcall = function(caller) {
 };
 ```
 
+=
+
 #### `admin`
 
 By default: `admin` is `false`:
@@ -100,27 +116,26 @@ By default: `admin` is `false`:
 call.admin = true;
 ```
 
-----
+=
 
 #### Demo
 
-1. https://webrtc-experiment.appspot.com/RTCall/
+1. https://www.webrtc-experiment.com/RTCall/
 
-----
+=
 
 #### Browser Support
 
-[RTCall.js](https://webrtc-experiment.appspot.com/RTCall/) supports following browsers:
+[RTCall.js](https://www.webrtc-experiment.com/RTCall/) supports following browsers:
 
 | Browser        | Support           |
 | ------------- |:-------------|
 | Firefox | [Stable](http://www.mozilla.org/en-US/firefox/new/) / [Aurora](http://www.mozilla.org/en-US/firefox/aurora/) / [Nightly](http://nightly.mozilla.org/) |
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
-| Internet Explorer / IE | [Chrome Frame](http://www.google.com/chromeframe) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[RTCall.js](https://webrtc-experiment.appspot.com/RTCall/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[RTCall.js](https://www.webrtc-experiment.com/RTCall/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/RTCall/index.html b/RTCall/index.html
index 478a77c0..e11df9ff 100644
--- a/RTCall/index.html
+++ b/RTCall/index.html
@@ -113,7 +113,7 @@
             padding-left: 4em;
             background-position: center left;
             color: white;
-            background-image: url(https://webrtc-experiment.appspot.com/images/call.png);
+            background-image: url(https://www.webrtc-experiment.com/images/call.png);
         }
 
             button:hover {
@@ -164,12 +164,12 @@
         document.createElement('footer');
     </script>
 
-    <script src="https://webrtc-experiment.appspot.com/RTCall.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
+    <script src="https://www.webrtc-experiment.com/RTCall.js"></script>
+    <script src="https://www.webrtc-experiment.com/firebase.js"></script>
 </head>
 
 <body>
-    <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none; right: 0; top: 0;">↑ WEBRTC EXPERIMENTS</a>
+    <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none; right: 0; top: 0;">↑ WEBRTC EXPERIMENTS</a>
 
     <article>
 
@@ -180,7 +180,7 @@ <h1>
         <div style="padding:1em;">
             Copyright © 2013 <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a>&lt;<a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>&gt;.
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
         </div>
 
@@ -352,7 +352,7 @@ <h2>Want to call admin?</h2>
         <h2>How to make audio-only calls using <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCall" target="_blank">RTCall</a>?</h2>
 
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/RTCall.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/RTCall.js"&gt;&lt;/script&gt;
 </pre>
 
         <pre>
@@ -404,7 +404,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> ©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> ©
             <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, 2013 »
             <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
             <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
@@ -412,6 +412,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         </p>
     </footer>
 
-    <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+    <script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/RecordRTC/README.md b/RecordRTC/README.md
index 517a43b5..d604d59a 100644
--- a/RecordRTC/README.md
+++ b/RecordRTC/README.md
@@ -1,6 +1,6 @@
-#### RecordRTC: WebRTC audio/video recording / [Demo](https://webrtc-experiment.appspot.com/RecordRTC/)
+#### RecordRTC: WebRTC audio/video recording / [Demo](https://www.webrtc-experiment.com/RecordRTC/)
 
-[RecordRTC](https://webrtc-experiment.appspot.com/RecordRTC.js) allows you record audio and/or video streams.
+[RecordRTC](https://www.webrtc-experiment.com/RecordRTC.js) allows you record audio and/or video streams.
 
 Note: For Mozilla Firefox; you can try [AudioVideoRecorder.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/AudioVideoRecorder).
 
@@ -22,7 +22,7 @@ Other features are:
 ##### How to use RecordRTC?
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/RecordRTC.js"></script>
+<script src="https://www.webrtc-experiment.com/RecordRTC.js"></script>
 ```
 
 =
@@ -48,7 +48,7 @@ recorder.stopVideo(function(videoURL) {
 ##### How to record animated GIF image?
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/gif-recorder.js"></script>
+<script src="https://www.webrtc-experiment.com/gif-recorder.js"></script>
 ```
 
 ```javascript
@@ -82,7 +82,7 @@ recorder.stopGIF(function(gifURL) {
 ##### How to record audio?
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/audio-recorder.js"></script>
+<script src="https://www.webrtc-experiment.com/audio-recorder.js"></script>
 ```
 
 ```javascript
@@ -236,7 +236,7 @@ Stereo audio is only supported for WAV files.
 
 ##### Browser Support
 
-[RecordRTC Demo](https://webrtc-experiment.appspot.com/RecordRTC/) works fine on following web-browsers:
+[RecordRTC Demo](https://www.webrtc-experiment.com/RecordRTC/) works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -259,4 +259,4 @@ Stereo audio is only supported for WAV files.
 
 ##### License
 
-[RecordRTC](https://webrtc-experiment.appspot.com/RecordRTC/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[RecordRTC](https://www.webrtc-experiment.com/RecordRTC/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/RecordRTC/index.html b/RecordRTC/index.html
index e01c4cb0..7bd14ef7 100644
--- a/RecordRTC/index.html
+++ b/RecordRTC/index.html
@@ -122,14 +122,14 @@
 
             ::-moz-selection { background: #ccc; }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/RecordRTC.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/gif-recorder.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/audio-recorder.js"> </script>
+        <script src="https://www.webrtc-experiment.com/RecordRTC.js"> </script>
+        <script src="https://www.webrtc-experiment.com/gif-recorder.js"> </script>
+        <script src="https://www.webrtc-experiment.com/audio-recorder.js"> </script>
     </head>
     <body>
         <h1>
             <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC" target="_blank">
-                RecordRTC</a>: <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC</a>
+                RecordRTC</a>: <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC</a>
             audio/video recording / <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC"
                                        target="_blank">Source Code</a>
         </h1>
@@ -137,7 +137,7 @@ <h1>
             Copyright © 2013 <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a>&lt;<a
                                                                                                          href="http://twitter.com/muazkh" target="_blank">@muazkh</a>&gt;.</p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <div>
             <h2>
@@ -311,7 +311,7 @@ <h2>
         <h2>
             How to use RecordRTC?</h2>
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/RecordRTC.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/RecordRTC.js"&gt;&lt;/script&gt;
 </pre>
         <h2>
             How to record video using RecordRTC?</h2>
@@ -331,7 +331,7 @@ <h2>
         <h2>
             How to record audio using RecordRTC?</h2>
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/audio-recorder.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/audio-recorder.js"&gt;&lt;/script&gt;
 var recorder = RecordRTC({
    stream: MediaStream || LocalMediaStream
 });
@@ -347,7 +347,7 @@ <h2>
         <h2>
             How to record animated GIF using RecordRTC?</h2>
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/gif-recorder.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/gif-recorder.js"&gt;&lt;/script&gt;
 var recorder = RecordRTC({
    video: HTMLVideoElement
 });
@@ -408,12 +408,12 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
         </section>
 
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span> 2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span> 2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                     href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-       <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+       <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
\ No newline at end of file
diff --git a/audio-broadcast/README.md b/audio-broadcast/README.md
index a129a82f..e056c3b0 100644
--- a/audio-broadcast/README.md
+++ b/audio-broadcast/README.md
@@ -1,11 +1,11 @@
-#### WebRTC One-to-Many audio sharing/broadcasting / [Demo](https://webrtc-experiment.appspot.com/audio-broadcast/)
+#### WebRTC One-to-Many audio sharing/broadcasting / [Demo](https://www.webrtc-experiment.com/audio-broadcast/)
 
 If 10 users join your broadcasted room, **20 RTP ports** will be opened on your browser:
 
 1. 10 RTP ports for **outgoing** audio streams
 2. 10 RTP ports for **incoming** audio streams
 
-----
+=
 
 #### Difference between one-way broadcasting and one-to-many broadcasting
 
@@ -15,11 +15,11 @@ On each participant's side; only one **incoming** RTP port will be opened.
 
 Unlike one-way broadcasting; one-to-many broadcasting experiment opens both outgoing as well as incoming RTP ports for each participant.
 
-----
+=
 
 #### Browser Support 
 
-This [WebRTC Audio Broadcasting Experiment](https://webrtc-experiment.appspot.com/audio-broadcast/) works fine on following web-browsers:
+This [WebRTC Audio Broadcasting Experiment](https://www.webrtc-experiment.com/audio-broadcast/) works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -27,8 +27,8 @@ This [WebRTC Audio Broadcasting Experiment](https://webrtc-experiment.appspot.co
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-WebRTC [Audio Broadcasting Experiment](https://webrtc-experiment.appspot.com/audio-broadcast/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+WebRTC [Audio Broadcasting Experiment](https://www.webrtc-experiment.com/audio-broadcast/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/audio-broadcast/index.html b/audio-broadcast/index.html
index 5fac1d17..687268ef 100644
--- a/audio-broadcast/index.html
+++ b/audio-broadcast/index.html
@@ -168,7 +168,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
         <h1>WebRTC one-to-many audio broadcasting / <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/audio-broadcast" target="_blank">Source Code</a></h1>
         <p>
@@ -177,7 +177,7 @@ <h1>WebRTC one-to-many audio broadcasting / <a href="https://github.com/muaz-kha
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table class="visible">
@@ -204,10 +204,10 @@ <h2>
 
         <div id="participants"></div>
 
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/audio-broadcast/broadcast.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/audio-broadcast/broadcast-ui.js"></script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
+        <script src="https://www.webrtc-experiment.com/audio-broadcast/broadcast.js"> </script>
+        <script src="https://www.webrtc-experiment.com/audio-broadcast/broadcast-ui.js"></script>
     </article>
 	<article>
 
@@ -235,13 +235,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/broadcast/README.md b/broadcast/README.md
index 617f4af2..f64b2f8e 100644
--- a/broadcast/README.md
+++ b/broadcast/README.md
@@ -1,6 +1,6 @@
-#### WebRTC One-to-Many video sharing/broadcasting / [Demo](https://webrtc-experiment.appspot.com/broadcast/)
+#### WebRTC One-to-Many video sharing/broadcasting / [Demo](https://www.webrtc-experiment.com/broadcast/)
 
-----
+=
 
 If 10 users join your broadcasted room, **40 RTP ports** will be opened on your browser:
 
@@ -9,7 +9,7 @@ If 10 users join your broadcasted room, **40 RTP ports** will be opened on your
 3. 10 RTP ports for **incoming** audio streams
 4. 10 RTP ports for **incoming** video streams
 
-----
+=
 
 #### Difference between one-way broadcasting and one-to-many broadcasting
 
@@ -24,11 +24,11 @@ On each participant's side; only 2 **incoming** RTP ports will be opened.
 
 Unlike one-way broadcasting; one-to-many broadcasting experiment opens both outgoing as well as incoming RTP ports for each participant.
 
-----
+=
 
 #### Browser Support 
 
-This [WebRTC Video Broadcasting Experiment](https://webrtc-experiment.appspot.com/broadcast/) works fine on following web-browsers:
+This [WebRTC Video Broadcasting Experiment](https://www.webrtc-experiment.com/broadcast/) works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -36,8 +36,8 @@ This [WebRTC Video Broadcasting Experiment](https://webrtc-experiment.appspot.co
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-WebRTC [Video Broadcasting Experiment](https://webrtc-experiment.appspot.com/broadcast/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+WebRTC [Video Broadcasting Experiment](https://www.webrtc-experiment.com/broadcast/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/broadcast/index.html b/broadcast/index.html
index fee8c57c..d07ecf21 100644
--- a/broadcast/index.html
+++ b/broadcast/index.html
@@ -172,7 +172,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
         <h1>WebRTC one-to-many video broadcasting / <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/broadcast" target="_blank">Source Code</a></h1>
         <p>
@@ -181,7 +181,7 @@ <h1>WebRTC one-to-many video broadcasting / <a href="https://github.com/muaz-kha
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table class="visible">
@@ -208,10 +208,10 @@ <h2>
 
         <div id="participants"></div>
 
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/broadcast/broadcast.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/broadcast/broadcast-ui.js"></script>		
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
+        <script src="https://www.webrtc-experiment.com/broadcast/broadcast.js"> </script>
+        <script src="https://www.webrtc-experiment.com/broadcast/broadcast-ui.js"></script>		
     </article>
     <article>
         <h2>Features</h2>
@@ -245,13 +245,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/chat-hangout/README.md b/chat-hangout/README.md
index 377c17f6..457ef041 100644
--- a/chat-hangout/README.md
+++ b/chat-hangout/README.md
@@ -1,4 +1,4 @@
-#### WebRTC P2P Group Text Chat / [Demo](https://webrtc-experiment.appspot.com/chat-hangout/)
+#### WebRTC P2P Group Text Chat / [Demo](https://www.webrtc-experiment.com/chat-hangout/)
 
 This WebRTC Experiment allows you share text messages among group of people.
 
@@ -15,7 +15,7 @@ So, `20` RTP data ports will be opened in `10` users data session. **Embarrassin
 
 On Firefox, by default 16 SCTP data ports will be opened for single peer. So, about 160 SCTP data ports will be opened in 10 users data session. Too awkward!
 
-----
+=
 
 #### Multiple peer connections.....is it a solution?
 
@@ -23,7 +23,7 @@ No, not at all. It is just a **temporary** workaround.
 
 You're strongly suggested to use **peer-to-server** model instead of opening multi-peers.
 
-----
+=
 
 #### How peer-to-server model works?
 
@@ -35,11 +35,11 @@ Remember, WebRTC peer object will send **DTLS/SRTP** packets maybe as **ByteStre
 
 Server can manipulate messages or data coming from 10 or more unique data ports and transfer over single data port!
 
-----
+=
 
 #### Browser Support
 
-WebRTC [Group Text Chat](https://webrtc-experiment.appspot.com/chat-hangout/) experiment works fine on following web-browsers:
+WebRTC [Group Text Chat](https://www.webrtc-experiment.com/chat-hangout/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -47,8 +47,8 @@ WebRTC [Group Text Chat](https://webrtc-experiment.appspot.com/chat-hangout/) ex
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-WebRTC [Group Text Chat](https://webrtc-experiment.appspot.com/chat-hangout/) experiment is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+WebRTC [Group Text Chat](https://www.webrtc-experiment.com/chat-hangout/) experiment is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/chat-hangout/hangout-ui.js b/chat-hangout/hangout-ui.js
index 5f4a67ca..8f7ab6f9 100644
--- a/chat-hangout/hangout-ui.js
+++ b/chat-hangout/hangout-ui.js
@@ -1,25 +1,36 @@
-/*
- 2013, @muazkh » github.com/muaz-khan
- MIT License » https://webrtc-experiment.appspot.com/licence/
- Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel
-*/
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/chat-hangout
 
 var config = {
-    openSocket: function (config) {
-        var channel = config.channel || location.hash.replace('#', '') || 'chat-hangout';
-        var socket = new Firebase('https://rtcweb.firebaseIO.com/' + channel);
+    openSocket: function(config) {
+        var SIGNALING_SERVER = 'https://www.webrtc-experiment.com:8553/',
+            defaultChannel = location.hash.substr(1) || 'group-text-chat-hangout';
+
+        var channel = config.channel || defaultChannel;
+        var sender = Math.round(Math.random() * 999999999) + 999999999;
+
+        io.connect(SIGNALING_SERVER).emit('new-channel', {
+            channel: channel,
+            sender: sender
+        });
+
+        var socket = io.connect(SIGNALING_SERVER + channel);
         socket.channel = channel;
-        socket.on('child_added', function (data) {
-            config.onmessage(data.val());
+        socket.on('connect', function() {
+            if (config.callback) config.callback(socket);
         });
-        socket.send = function (data) {
-            this.push(data);
-        }
-        config.onopen && setTimeout(config.onopen, 1);
-        socket.onDisconnect().remove();
-        return socket;
+
+        socket.send = function(message) {
+            socket.emit('message', {
+                sender: sender,
+                data: message
+            });
+        };
+
+        socket.on('message', config.onmessage);
     },
-    onRoomFound: function (room) {
+    onRoomFound: function(room) {
         var alreadyExist = document.getElementById(room.broadcaster);
         if (alreadyExist) return;
 
@@ -28,11 +39,11 @@ var config = {
         var tr = document.createElement('tr');
         tr.setAttribute('id', room.broadcaster);
         tr.innerHTML = '<td>' + room.roomName + '</td>' +
-            '<td><button class="join" id="' + room.roomToken + '">Join Room</button></td>';
+            '<td><button class="join" id="' + room.roomToken + '">Join</button></td>';
 
         roomsList.insertBefore(tr, roomsList.firstChild);
 
-        tr.onclick = function () {
+        tr.onclick = function() {
             var tr = this;
             hangoutUI.joinRoom({
                 roomToken: tr.querySelector('.join').id,
@@ -42,11 +53,10 @@ var config = {
             hideUnnecessaryStuff();
         };
     },
-    onChannelOpened: function (/* channel */) {
+    onChannelOpened: function(/* channel */) {
         hideUnnecessaryStuff();
     },
-    onChannelMessage: function (data) {
-        console.log(data);
+    onChannelMessage: function(data) {
         if (!chatOutput) return;
 
         var tr = document.createElement('tr');
@@ -61,7 +71,7 @@ var config = {
 function createButtonClickHandler() {
     hangoutUI.createRoom({
         userName: prompt('Enter your name', 'Anonymous'),
-        roomName: ((document.getElementById('conference-name') || { value: null }).value || 'Anonymous') + ' // shared via ' + (navigator.vendor ? 'Google Chrome (Stable/Canary)' : 'Mozilla Firefox (Aurora/Nightly)')
+        roomName: ((document.getElementById('conference-name') || { }).value || 'Anonymous') + ' // shared via ' + (!!navigator.webkitGetUserMedia ? 'Google Chrome (Stable/Canary)' : 'Mozilla Firefox (Aurora/Nightly)')
     });
     hideUnnecessaryStuff();
 }
@@ -93,7 +103,7 @@ function hideUnnecessaryStuff() {
 
 var chatMessage = document.getElementById('chat-message');
 if (chatMessage)
-    chatMessage.onchange = function () {
+    chatMessage.onchange = function() {
         hangoutUI.send(this.value);
         var tr = document.createElement('tr');
         tr.innerHTML =
@@ -105,8 +115,9 @@ if (chatMessage)
     };
 
 
-(function () {
+(function() {
     var uniqueToken = document.getElementById('unique-token');
-    if (uniqueToken) if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
-    else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace(/\./g, '-');
-})();
\ No newline at end of file
+    if (uniqueToken)
+        if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
+        else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace( /\./g , '-');
+})();
diff --git a/chat-hangout/hangout.js b/chat-hangout/hangout.js
index a5c5e0b5..a807d5c5 100644
--- a/chat-hangout/hangout.js
+++ b/chat-hangout/hangout.js
@@ -1,6 +1,8 @@
-/* MIT License: https://webrtc-experiment.appspot.com/licence/  */
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/chat-hangout
 
-var hangout = function (config) {
+var hangout = function(config) {
     var self = {
         userToken: uniqueToken(),
         userName: 'Anonymous'
@@ -9,15 +11,15 @@ var hangout = function (config) {
         isbroadcaster,
         isGetNewRoom = true,
         sockets = [],
-        defaultSocket = {}, RTCDataChannels = [];
+        defaultSocket = { }, RTCDataChannels = [];
 
     function openDefaultSocket() {
         defaultSocket = config.openSocket({
-                onmessage: onDefaultSocketResponse,
-                callback: function (socket) {
-                    defaultSocket = socket;
-                }
-            });
+            onmessage: onDefaultSocketResponse,
+            callback: function(socket) {
+                defaultSocket = socket;
+            }
+        });
     }
 
     function onDefaultSocketResponse(response) {
@@ -30,10 +32,10 @@ var hangout = function (config) {
         if (response.userToken && response.joinUser == self.userToken && response.participant && channels.indexOf(response.userToken) == -1) {
             channels += response.userToken + '--';
             openSubSocket({
-                    isofferer: true,
-                    channel: response.channel || response.userToken,
-                    closeSocket: true
-                });
+                isofferer: true,
+                channel: response.channel || response.userToken,
+                closeSocket: true
+            });
         }
     }
 
@@ -42,13 +44,13 @@ var hangout = function (config) {
         var socketConfig = {
             channel: _config.channel,
             onmessage: socketResponse,
-            onopen: function () {
+            onopen: function() {
                 if (isofferer && !peer) initPeer();
                 sockets[sockets.length] = socket;
             }
         };
 
-        socketConfig.callback = function (_socket) {
+        socketConfig.callback = function(_socket) {
             socket = _socket;
             this.onopen();
         };
@@ -56,21 +58,21 @@ var hangout = function (config) {
         var socket = config.openSocket(socketConfig),
             isofferer = _config.isofferer,
             gotstream,
-            inner = {},
+            inner = { },
             peer;
 
         var peerConfig = {
-            onICE: function (candidate) {
+            onICE: function(candidate) {
                 socket.send({
-                        userToken: self.userToken,
-                        candidate: {
-                            sdpMLineIndex: candidate.sdpMLineIndex,
-                            candidate: JSON.stringify(candidate.candidate)
-                        }
-                    });
+                    userToken: self.userToken,
+                    candidate: {
+                        sdpMLineIndex: candidate.sdpMLineIndex,
+                        candidate: JSON.stringify(candidate.candidate)
+                    }
+                });
             },
             onChannelOpened: onChannelOpened,
-            onChannelMessage: function (event) {
+            onChannelMessage: function(event) {
                 config.onChannelMessage(JSON.parse(event.data));
             }
         };
@@ -89,76 +91,43 @@ var hangout = function (config) {
         function onChannelOpened(channel) {
             RTCDataChannels[RTCDataChannels.length] = channel;
             channel.send(JSON.stringify({
-                        message: 'Hi, I\'m <strong>' + self.userName + '</strong>!',
-                        sender: self.userName
-                    }));
+                message: 'Hi, I\'m <strong>' + self.userName + '</strong>!',
+                sender: self.userName
+            }));
 
             if (config.onChannelOpened) config.onChannelOpened(channel);
 
             if (isbroadcaster && channels.split('--').length > 3) {
                 /* broadcasting newly connected participant for video-conferencing! */
                 defaultSocket.send({
-                        newParticipant: socket.channel,
-                        userToken: self.userToken
-                    });
+                    newParticipant: socket.channel,
+                    userToken: self.userToken
+                });
             }
 
             gotstream = true;
         }
 
         function sendsdp(sdp) {
-            sdp = JSON.stringify(sdp);
-            var part = parseInt(sdp.length / 3);
-
-            var firstPart = sdp.slice(0, part),
-                secondPart = sdp.slice(part, sdp.length - 1),
-                thirdPart = '';
-
-            if (sdp.length > part + part) {
-                secondPart = sdp.slice(part, part + part);
-                thirdPart = sdp.slice(part + part, sdp.length);
-            }
-
             socket.send({
-                    userToken: self.userToken,
-                    firstPart: firstPart
-                });
-
-            socket.send({
-                    userToken: self.userToken,
-                    secondPart: secondPart
-                });
-
-            socket.send({
-                    userToken: self.userToken,
-                    thirdPart: thirdPart
-                });
+                userToken: self.userToken,
+                sdp: JSON.stringify(sdp)
+            });
         }
 
         function socketResponse(response) {
             if (response.userToken == self.userToken) return;
 
-            if (response.firstPart || response.secondPart || response.thirdPart) {
-                if (response.firstPart) {
-                    inner.firstPart = response.firstPart;
-                    if (inner.secondPart && inner.thirdPart) selfInvoker();
-                }
-                if (response.secondPart) {
-                    inner.secondPart = response.secondPart;
-                    if (inner.firstPart && inner.thirdPart) selfInvoker();
-                }
-
-                if (response.thirdPart) {
-                    inner.thirdPart = response.thirdPart;
-                    if (inner.firstPart && inner.secondPart) selfInvoker();
-                }
+            if (response.sdp) {
+                inner.sdp = JSON.parse(response.sdp);
+                selfInvoker();
             }
 
             if (response.candidate && !gotstream) {
                 peer && peer.addICE({
-                        sdpMLineIndex: response.candidate.sdpMLineIndex,
-                        candidate: JSON.parse(response.candidate.candidate)
-                    });
+                    sdpMLineIndex: response.candidate.sdpMLineIndex,
+                    candidate: JSON.parse(response.candidate.candidate)
+                });
             }
 
             if (response.left) {
@@ -176,53 +145,39 @@ var hangout = function (config) {
 
             invokedOnce = true;
 
-            inner.sdp = JSON.parse(inner.firstPart + inner.secondPart + inner.thirdPart);
-
             if (isofferer) peer.addAnswerSDP(inner.sdp);
             else initPeer(inner.sdp);
         }
     }
 
     function leave() {
-        length = sockets.length;
+        var length = sockets.length;
         for (var i = 0; i < length; i++) {
-            socket = sockets[i];
+            var socket = sockets[i];
             if (socket) {
                 socket.send({
-                        left: true,
-                        userToken: self.userToken
-                    });
+                    left: true,
+                    userToken: self.userToken
+                });
                 delete sockets[i];
             }
         }
     }
 
-    window.onunload = function () {
+    window.onbeforeunload = function() {
         leave();
     };
 
-    window.onkeyup = function (e) {
+    window.onkeyup = function(e) {
         if (e.keyCode == 116) leave();
     };
 
-    (function () {
-            var anchors = document.querySelectorAll('a'),
-                length = anchors.length;
-            for (var i = 0; i < length; i++) {
-                a = anchors[i];
-                if (a.href.indexOf('#') !== 0 && a.getAttribute('target') != '_blank')
-                    a.onclick = function () {
-                        leave();
-                };
-            }
-        })();
-
     function startBroadcasting() {
         defaultSocket && defaultSocket.send({
-                roomToken: self.roomToken,
-                roomName: self.roomName,
-                broadcaster: self.userToken
-            });
+            roomToken: self.roomToken,
+            roomName: self.roomName,
+            broadcaster: self.userToken
+        });
         setTimeout(startBroadcasting, 3000);
     }
 
@@ -232,20 +187,20 @@ var hangout = function (config) {
 
         var new_channel = uniqueToken();
         openSubSocket({
-                channel: new_channel,
-                closeSocket: true
-            });
+            channel: new_channel,
+            closeSocket: true
+        });
 
         defaultSocket.send({
-                participant: true,
-                userToken: self.userToken,
-                joinUser: channel,
-                channel: new_channel
-            });
+            participant: true,
+            userToken: self.userToken,
+            joinUser: channel,
+            channel: new_channel
+        });
     }
 
     function uniqueToken() {
-        var s4 = function () {
+        var s4 = function() {
             return Math.floor(Math.random() * 0x10000).toString(16);
         };
         return s4() + s4() + "-" + s4() + "-" + s4() + "-" + s4() + "-" + s4() + s4() + s4();
@@ -253,7 +208,7 @@ var hangout = function (config) {
 
     openDefaultSocket();
     return {
-        createRoom: function (_config) {
+        createRoom: function(_config) {
             self.roomName = _config.roomName || 'Anonymous';
             self.roomToken = uniqueToken();
             if (_config.userName) self.userName = _config.userName;
@@ -262,31 +217,33 @@ var hangout = function (config) {
             isGetNewRoom = false;
             startBroadcasting();
         },
-        joinRoom: function (_config) {
+        joinRoom: function(_config) {
             self.roomToken = _config.roomToken;
             if (_config.userName) self.userName = _config.userName;
             isGetNewRoom = false;
 
             openSubSocket({
-                    channel: self.userToken
-                });
+                channel: self.userToken
+            });
 
             defaultSocket.send({
-                    participant: true,
-                    userToken: self.userToken,
-                    joinUser: _config.joinUser
-                });
+                participant: true,
+                userToken: self.userToken,
+                joinUser: _config.joinUser
+            });
         },
-        send: function (message) {
-            console.log('list of data channels', RTCDataChannels);
+        send: function(message) {
             var length = RTCDataChannels.length,
                 data = JSON.stringify({
-                        message: message,
-                        sender: self.userName
-                    });
+                    message: message,
+                    sender: self.userName
+                });
             if (!length) return;
             for (var i = 0; i < length; i++) {
-                RTCDataChannels[i].send(data);
+                try {
+                    RTCDataChannels[i].send(data);
+                } catch(e) {
+                }
             }
         }
     };
diff --git a/chat-hangout/index.html b/chat-hangout/index.html
index 7acabf92..83ed72b2 100644
--- a/chat-hangout/index.html
+++ b/chat-hangout/index.html
@@ -178,7 +178,7 @@
 
 <body>
 <article>
-    <a href="https://webrtc-experiment.appspot.com/"
+    <a href="https://www.webrtc-experiment.com/"
        style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
         WEBRTC EXPERIMENTS</a>
 
@@ -191,7 +191,7 @@ <h1><a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/chat-han
     </p>
 
     <section class="plusone-gplus">
-        <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+        <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
     </section>
 
     <!-- <required> -->
@@ -228,10 +228,10 @@ <h1><a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/chat-han
     </table>
     <table id="chat-output" class="hidden"></table>
 
-    <script src='https://webrtc-experiment.appspot.com/firebase.js'></script>
-    <script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/chat-hangout/hangout.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/chat-hangout/hangout-ui.js"></script>
+    <script src="https://www.webrtc-experiment.com/socket.io.js"></script>
+    <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
+    <script src="https://www.webrtc-experiment.com/chat-hangout/hangout.js"></script>
+    <script src="https://www.webrtc-experiment.com/chat-hangout/hangout-ui.js"></script>
 
     <br/>
     <br/>
@@ -256,7 +256,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -264,6 +264,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/chat/README.md b/chat/README.md
index 65125a4b..ebb66e82 100644
--- a/chat/README.md
+++ b/chat/README.md
@@ -1,14 +1,18 @@
-#### WebRTC P2P Text Chat / [Demo](https://webrtc-experiment.appspot.com/chat-hangout/)
+#### WebRTC P2P Text Chat / [Demo](https://www.webrtc-experiment.com/chat-hangout/)
 
 This WebRTC Experiment allows you share text messages among many peers.
 
 It opens multiple peer connections to support group data broadcasting.
 
+=
+
 #### On room initiator's side
 
 1. 10 peer connections will be opened
 2. 10 sockets will be opened to exchange SDP/ICE
 
+=
+
 #### On chrome, how many RTP data ports will be opened
 
 In 10 users data session; about 20 RTP data ports will be opened:
@@ -20,10 +24,14 @@ Firefox opens 16 SCTP data ports for single data session.
 
 It means that about 160 SCTP data ports will be opened on room initiator's side in 10 users data session.
 
+=
+
 #### On participants' side
 
 Only 2 RTP data ports will be opened because participants are not connected with each other. They're connected directly with room initiator.
 
+=
+
 #### License
 
-[WebRTC Experiments](https://webrtc-experiment.appspot.com/) are released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC Experiments](https://www.webrtc-experiment.com/) are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/demos/README.md b/demos/README.md
index 9167e1df..ffcc1e98 100644
--- a/demos/README.md
+++ b/demos/README.md
@@ -2,6 +2,8 @@
 
 Plug & play type of WebRTC Experiments. Nothing to install. No requirements. Just copy JavaScript code in your site and that's all you need to do!
 
+=
+
 #### Chrome-to-Chrome WebRTC Data Connection
 
 It is a single-page demo. Copy/paste code in the console tab and you're done!
@@ -85,6 +87,8 @@ offererDataChannel.send('text message');
 answererDataChannel.send('text message');
 ```
 
+=
+
 #### Firefox-to-Firefox WebRTC Data Connection
 
 From April 2013, Mozilla did [broken changes](https://github.com/muaz-khan/WebRTC-Experiment/wiki/WebRTC-DataChannel-and-Firefox#points) in Firefox data channel implementation. So, following code will work with upcoming releases.
@@ -258,9 +262,11 @@ function setChannelEvents(channel, channelNameForConsoleOutput) {
 function useless() { }
 ```
 
+=
+
 #### Browser Support
 
-[WebRTC Experiments & Demos](https://webrtc-experiment.appspot.com) works fine on following web-browsers:
+[WebRTC Experiments & Demos](https://www.webrtc-experiment.com) works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -268,13 +274,15 @@ function useless() { }
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Internet Explorer / IE | [Chrome Frame](http://www.google.com/chromeframe) |
 
-| [WebRTC One-Page Demos](https://webrtc-experiment.appspot.com)        |
+| [WebRTC One-Page Demos](https://www.webrtc-experiment.com)        |
 | ------------- |
-| [One-Page text chat using RTCDataChannel APIs](https://webrtc-experiment.appspot.com/demos/client-side-datachannel.html) |
-| [One-Page direct video chat](https://webrtc-experiment.appspot.com/demos/client-side.html) |
-| [One-Page video chat using socket.io for signaling](https://webrtc-experiment.appspot.com/demos/client-side-socket-io.html) |
-| [One-Page video chat using WebSockets for signaling](https://webrtc-experiment.appspot.com/demos/client-side-websocket.html) |
+| [One-Page text chat using RTCDataChannel APIs](https://www.webrtc-experiment.com/demos/client-side-datachannel.html) |
+| [One-Page direct video chat](https://www.webrtc-experiment.com/demos/client-side.html) |
+| [One-Page video chat using socket.io for signaling](https://www.webrtc-experiment.com/demos/client-side-socket-io.html) |
+| [One-Page video chat using WebSockets for signaling](https://www.webrtc-experiment.com/demos/client-side-websocket.html) |
+
+=
 
 #### License
 
-[WebRTC Experiments & Demos](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC Experiments & Demos](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/demos/client-side-datachannel.html b/demos/client-side-datachannel.html
index 30167c2c..94936c3f 100644
--- a/demos/client-side-datachannel.html
+++ b/demos/client-side-datachannel.html
@@ -178,7 +178,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
         <h1>RTCDataChannel Simple Demo
             / <a href="https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos" target="_blank">Source Code on Github</a>
         </h1>
@@ -188,7 +188,7 @@ <h1>RTCDataChannel Simple Demo
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table>
@@ -403,11 +403,11 @@ <h1>RTCDataChannel Simple Demo
         <br />
 
         <h2>
-            <a href="https://webrtc-experiment.appspot.com/docs/how-to-use-rtcdatachannel.html" target="_blank">How to use RTCDataChannel?</a>
+            <a href="https://www.webrtc-experiment.com/docs/how-to-use-rtcdatachannel.html" target="_blank">How to use RTCDataChannel?</a>
         </h2>
         :
         <h2>
-            <a href="https://webrtc-experiment.appspot.com/docs/rtc-datachannel-for-beginners.html" target="_blank">RTCDataChannel for Beginners!</a>
+            <a href="https://www.webrtc-experiment.com/docs/rtc-datachannel-for-beginners.html" target="_blank">RTCDataChannel for Beginners!</a>
         </h2>
 
         <br />
@@ -422,13 +422,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/demos/client-side.html b/demos/client-side.html
index a00a9f26..71973dd6 100644
--- a/demos/client-side.html
+++ b/demos/client-side.html
@@ -158,7 +158,7 @@
     </head>
     
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
 
              <h1>RTCPeerConnection Simple Demo
             / <a href="https://github.com/muaz-khan/WebRTC-Experiment/blob/master/demos" target="_blank">Source Code on Github</a>
@@ -169,7 +169,7 @@ <h1>RTCPeerConnection Simple Demo
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <div style="text-align:center;">
                 <div class="videos-container">
@@ -311,13 +311,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                 href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/demos/remote-stream-recording.html b/demos/remote-stream-recording.html
new file mode 100644
index 00000000..0791d132
--- /dev/null
+++ b/demos/remote-stream-recording.html
@@ -0,0 +1,171 @@
+<title>Recording Remote Audio Streams / RecordRTC</title>
+<link href='https://fonts.googleapis.com/css?family=Open+Sans' rel='stylesheet' type='text/css'>
+<style>
+html{
+	background: rgb(231, 241, 241);
+	font-family: 'Open Sans', sans-serif;
+    font-size: 1.2em;
+    line-height: 1.5;
+}
+h1 {
+	text-align: center;
+	border-bottom: 2px solid rgb(121, 40, 40);
+	color: rgb(121, 40, 40);
+}
+ol {
+    margin-left: 4em;
+}
+pre {
+    margin-left: 6em;
+}
+</style>
+<h1>Recording Remote Audio Streams / <a href="https://www.webrtc-experiment.com/RecordRTC/">RecordRTC</a></h2>
+
+<h2>issue: <a href="http://stackoverflow.com/questions/17475038/recording-a-remote-webrtc-stream-with-recordrtc/17475416#17475416" target="_blank">unable to record remote audio streams using RecordRTC.</a></h2>
+<blockquote>
+"opus" is the default codec on chrome;
+</blockquote>
+
+<blockquote>
+Chrome uses two modes for opus....
+</blockquote>
+
+<ol>
+<li>"voip" (i.e. voice mode) which is used if mono audio stream is provided by the  sender</li>
+<li>"audio" which is used if stereo audio stream is provided by  the sender</li>
+</ol>
+<blockquote>
+There is no parameter in sdp allows us control voip/audio mode for outgoing/incoming audio streams. This job can be done by sender ( I don't know how, well, still experimenting ).
+</blockquote>
+
+<blockquote>
+They're using code like this:
+</blockquote>
+
+<pre>
+// Default to VoIP application for mono, and AUDIO for stereo.
+int application = (channels == 1) ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO;
+
+state->encoder = opus_encoder_create(48000, channels, application, &error);
+</pre>
+
+<blockquote>
+You can see that if channels equals 1; i.e. if mono audio; they're using "voip" mode. Otherwise "audio" mode.
+</blockquote>
+
+<blockquote>
+In the session description; you can see this line:
+</blockquote>
+
+<pre>
+a=rtpmap:111 opus/48000/2
+</pre>
+
+<blockquote>
+It is mono section with 48000 Hz (i.e. highest) clock rate.
+</blockquote>
+
+<pre>
+Bitrate is 48-64 kb/s .... i.e. FB mono music
+</pre>
+
+<blockquote>
+For stereo; we need:
+</blockquote>
+
+<pre>
+64-128 kb/s .... i.e. FB stereo music
+</pre>
+
+<script src="https://www.webrtc-experiment.com/RecordRTC.js"></script>
+
+<script>
+    var offerer, answerer;
+
+    window.RTCPeerConnection = window.mozRTCPeerConnection || window.webkitRTCPeerConnection;
+    window.RTCSessionDescription = window.mozRTCSessionDescription || window.RTCSessionDescription;
+    window.RTCIceCandidate = window.mozRTCIceCandidate || window.RTCIceCandidate;
+
+    navigator.getUserMedia = navigator.mozGetUserMedia || navigator.webkitGetUserMedia;
+    window.URL = window.webkitURL || window.URL;
+
+    window.iceServers = {
+        iceServers: [{
+                url: 'stun:23.21.150.121'
+            }
+        ]
+    };
+</script>
+<script>
+    /* offerer */
+
+    function offererPeer(stream) {
+        offerer = new RTCPeerConnection(window.iceServers);
+
+        // stream is only attached by offerer
+        offerer.addStream(stream);
+
+        offerer.onicecandidate = function (event) {
+            if (!event || !event.candidate) return;
+            answerer.addIceCandidate(event.candidate);
+        };
+
+        offerer.createOffer(function (offer) {
+            console.debug('offer sdp', offer.sdp);
+
+            offerer.setLocalDescription(offer);
+            answererPeer(offer);
+        });
+    }
+</script>
+<script>
+    /* answerer */
+
+    function answererPeer(offer) {
+        answerer = new RTCPeerConnection(window.iceServers);
+
+        // stream that is flowing from offerer toward answerer
+        answerer.onaddstream = function (event) {
+            console.debug('remote stream', event.stream);
+			
+			var recorder = new RecordRTC({
+				stream: event.stream
+			});
+			recorder.recordAudio();
+			setTimeout(function() {
+				recorder.stopAudio(function() {
+					var dataURL = recorder.getDataURL();
+					if(dataURL.indexOf('AAAAAAAAAAAA') != -1) {
+						document.querySelector('h2').innerHTML = 'Unable to record remote audio stream.<br /><br />' +
+																  dataURL.substr(0, 150) + '...';
+					}
+					console.log(dataURL);
+				});
+			}, 5000);
+        };
+
+        answerer.onicecandidate = function (event) {
+            if (!event || !event.candidate) return;
+            offerer.addIceCandidate(event.candidate);
+        };
+
+        answerer.setRemoteDescription(offer);
+        answerer.createAnswer(function (answer) {
+            answerer.setLocalDescription(answer);
+            offerer.setRemoteDescription(answer);
+        });
+    }
+</script>
+<script>
+    function getUserMedia(callback) {
+        navigator.getUserMedia({
+            audio: true,
+            video: false
+        }, callback, onerror);
+
+        function onerror(e) {
+            console.error(e);
+        }
+    }
+    getUserMedia(offererPeer);
+</script>
\ No newline at end of file
diff --git a/demos/screen-and-video-from-single-peer.html b/demos/screen-and-video-from-single-peer.html
index 89c69cb3..9a0a0b1a 100644
--- a/demos/screen-and-video-from-single-peer.html
+++ b/demos/screen-and-video-from-single-peer.html
@@ -166,7 +166,7 @@
         </script>
     </head>
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
 
             <h1>Share screen and audio/video from single peer connection!</h1>
 
@@ -175,7 +175,7 @@ <h1>Share screen and audio/video from single peer connection!</h1>
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <div style="text-align: center;">
                 <div class="videos-container">
@@ -352,12 +352,12 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                     href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
\ No newline at end of file
diff --git a/docs/How-to-Broadcast-Screen-using-WebRTC.html b/docs/How-to-Broadcast-Screen-using-WebRTC.html
index e3edd280..70bc19da 100644
--- a/docs/How-to-Broadcast-Screen-using-WebRTC.html
+++ b/docs/How-to-Broadcast-Screen-using-WebRTC.html
@@ -166,7 +166,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
         <h1>How to share tab using tabCapture APIs?
         </h1>
         <p>
@@ -175,7 +175,7 @@ <h1>How to share tab using tabCapture APIs?
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         If you're newcomer, newbie or beginner; you're suggested to try <a href="http://bit.ly/RTCMultiConnection-Documentation" target="_blank">RTCMultiConnection.js</a> or <a href="http://bit.ly/DataChannel-Documentation" target="_blank">DataChannel.js</a> libraries.
@@ -338,7 +338,7 @@ <h2>To use your own socket.io implementation...</h2>
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
diff --git a/docs/README.md b/docs/README.md
index 19edd707..b692bda6 100644
--- a/docs/README.md
+++ b/docs/README.md
@@ -1,18 +1,21 @@
 | A few documents for newbies and beginners        |
 | ------------- |
 | [RTCPeerConnection Documentation](https://github.com/muaz-khan/WebRTC-Experiment/wiki/RTCPeerConnection-Documentation) |
-| [How to use RTCPeerConnection.js?](https://webrtc-experiment.appspot.com/docs/how-to-use-rtcpeerconnection-js-v1.1.html) |
-| [RTCDataChannel for Beginners](https://webrtc-experiment.appspot.com/docs/rtc-datachannel-for-beginners.html) |
-| [How to use RTCDataChannel?](https://webrtc-experiment.appspot.com/docs/how-to-use-rtcdatachannel.html) - single code for both canary and nightly |
-| [WebRTC for Beginners: A getting stared guide!](https://webrtc-experiment.appspot.com/docs/webrtc-for-beginners.html) |
-| [WebRTC for Newbies ](https://webrtc-experiment.appspot.com/docs/webrtc-for-newbies.html) |
-| [How to broadcast video using RTCWeb APIs?](https://webrtc-experiment.appspot.com/docs/how-to-broadcast-video-using-RTCWeb-APIs.html) |
-| [How to share audio-only streams?](https://webrtc-experiment.appspot.com/docs/how-to-share-audio-only-streams.html) |
-| [How to broadcast files using RTCDataChannel APIs?](https://webrtc-experiment.appspot.com/docs/how-file-broadcast-works.html) |
+| [How to use RTCPeerConnection.js?](https://www.webrtc-experiment.com/docs/how-to-use-rtcpeerconnection-js-v1.1.html) |
+| [RTCDataChannel for Beginners](https://www.webrtc-experiment.com/docs/rtc-datachannel-for-beginners.html) |
+| [How to use RTCDataChannel?](https://www.webrtc-experiment.com/docs/how-to-use-rtcdatachannel.html) - single code for both canary and nightly |
+| [WebRTC for Beginners: A getting stared guide!](https://www.webrtc-experiment.com/docs/webrtc-for-beginners.html) |
+| [WebRTC for Newbies ](https://www.webrtc-experiment.com/docs/webrtc-for-newbies.html) |
+| [How to broadcast video using RTCWeb APIs?](https://www.webrtc-experiment.com/docs/how-to-broadcast-video-using-RTCWeb-APIs.html) |
+| [How to share audio-only streams?](https://www.webrtc-experiment.com/docs/how-to-share-audio-only-streams.html) |
+| [How to broadcast files using RTCDataChannel APIs?](https://www.webrtc-experiment.com/docs/how-file-broadcast-works.html) |
 
-## Some other documents on [WebRTC Wiki Pages](https://github.com/muaz-khan/WebRTC-Experiment/wiki)
+=
 
-====
-## License
+##### Some other documents on [WebRTC Wiki Pages](https://github.com/muaz-khan/WebRTC-Experiment/wiki)
 
-[WebRTC Experiments](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+=
+
+##### License
+
+[WebRTC Experiments](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/docs/how-file-broadcast-works.html b/docs/how-file-broadcast-works.html
index 1ff61508..e073240e 100644
--- a/docs/how-file-broadcast-works.html
+++ b/docs/how-file-broadcast-works.html
@@ -151,9 +151,9 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
-        <h1>How to <a href="https://webrtc-experiment.appspot.com/file-hangout/" target="_blank">share files</a> using <a href="https://webrtc-experiment.appspot.com/docs/rtc-datachannel-for-beginners.html" target="_blank">RTCDataChannel API</a>s?
+        <h1>How to <a href="https://www.webrtc-experiment.com/file-hangout/" target="_blank">share files</a> using <a href="https://www.webrtc-experiment.com/docs/rtc-datachannel-for-beginners.html" target="_blank">RTCDataChannel API</a>s?
         </h1>
         <p>
             <span>Copyright © 2013</span>
@@ -161,7 +161,7 @@ <h1>How to <a href="https://webrtc-experiment.appspot.com/file-hangout/" target=
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <h2>Getting started — You'll</h2>
@@ -318,11 +318,11 @@ <h2>Links</h2>
 
         <ol>
             <li>
-                <a href="https://webrtc-experiment.appspot.com/file-hangout/" target="_blank">Share Files / Demo</a>
+                <a href="https://www.webrtc-experiment.com/file-hangout/" target="_blank">Share Files / Demo</a>
             </li>
 
             <li>
-                <a href="https://webrtc-experiment.appspot.com/docs/rtc-datachannel-for-beginners.html" target="_blank">RTCDataChannel APIs for Beginners</a>
+                <a href="https://www.webrtc-experiment.com/docs/rtc-datachannel-for-beginners.html" target="_blank">RTCDataChannel APIs for Beginners</a>
             </li>
 
             <li>
@@ -346,13 +346,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/docs/how-to-WebRTC-video-conferencing.html b/docs/how-to-WebRTC-video-conferencing.html
index 6850b49f..4f0b4512 100644
--- a/docs/how-to-WebRTC-video-conferencing.html
+++ b/docs/how-to-WebRTC-video-conferencing.html
@@ -172,7 +172,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
         <h1>How to write video-conferencing application using WebRTC?
         </h1>
         <p>
@@ -181,7 +181,7 @@ <h1>How to write video-conferencing application using WebRTC?
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table style="width: 100%;">
@@ -412,13 +412,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/docs/how-to-broadcast-video-using-RTCWeb-APIs.html b/docs/how-to-broadcast-video-using-RTCWeb-APIs.html
index 846eef45..94625b96 100644
--- a/docs/how-to-broadcast-video-using-RTCWeb-APIs.html
+++ b/docs/how-to-broadcast-video-using-RTCWeb-APIs.html
@@ -166,7 +166,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
         <h1>How to broadcast video using RTCWeb APIs?
         </h1>
@@ -176,7 +176,7 @@ <h1>How to broadcast video using RTCWeb APIs?
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table style="width: 100%;">
@@ -330,13 +330,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/docs/how-to-install-tabCapture-extension.html b/docs/how-to-install-tabCapture-extension.html
index 2748c78c..3cfba770 100644
--- a/docs/how-to-install-tabCapture-extension.html
+++ b/docs/how-to-install-tabCapture-extension.html
@@ -183,7 +183,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
         <h1>How to install tabCapture extension?</h1>
         <p>
             <span>Copyright © 2013</span>
@@ -191,7 +191,7 @@ <h1>How to install tabCapture extension?</h1>
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table>
@@ -216,13 +216,13 @@ <h1>How to install tabCapture extension?</h1>
             </tr>
 
             <tr>
-                <td>After broadcasting, you can see your tab is <a href="https://webrtc-experiment.appspot.com/screen-broadcast/">listed here</a>.
+                <td>After broadcasting, you can see your tab is <a href="https://www.webrtc-experiment.com/screen-broadcast/">listed here</a>.
                 </td>
             </tr>
 
             <tr>
                 <td>
-                    <img src="https://webrtc-experiment.appspot.com/images/screen-broadcasting.png" alt="Tab Broadcasting using WebRTC!" style="width: 100%;">
+                    <img src="https://www.webrtc-experiment.com/images/screen-broadcasting.png" alt="Tab Broadcasting using WebRTC!" style="width: 100%;">
                 </td>
             </tr>
         </table>
@@ -245,7 +245,7 @@ <h2><a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/webrtc-e
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
diff --git a/docs/how-to-share-audio-only-streams.html b/docs/how-to-share-audio-only-streams.html
index ed5c5f7c..aac9befa 100644
--- a/docs/how-to-share-audio-only-streams.html
+++ b/docs/how-to-share-audio-only-streams.html
@@ -166,7 +166,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
         <h1>How to share audio-only streams using RTCWeb APIs?
         </h1>
@@ -176,7 +176,7 @@ <h1>How to share audio-only streams using RTCWeb APIs?
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table style="width: 100%;">
@@ -235,13 +235,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/docs/how-to-use-plugin-free-calls.html b/docs/how-to-use-plugin-free-calls.html
index 12d1979c..204a4961 100644
--- a/docs/how-to-use-plugin-free-calls.html
+++ b/docs/how-to-use-plugin-free-calls.html
@@ -224,7 +224,7 @@
     </style>
 </head>
 <body>
-<a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+<a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
 
 <table>
@@ -298,7 +298,7 @@ <h1>How to use <a href="/calls/">Plugin-free calls</a>?</h1>
         </tr>
     </table>
     <section class="plusone-gplus">
-        <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+        <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
     </section>
     <footer>
         <a href="/">Home</a>
@@ -309,6 +309,6 @@ <h1>How to use <a href="/calls/">Plugin-free calls</a>?</h1>
     </footer>
 
 
-    <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+    <script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
diff --git a/docs/how-to-use-rtcdatachannel.html b/docs/how-to-use-rtcdatachannel.html
index 3de397f6..ac07e31a 100644
--- a/docs/how-to-use-rtcdatachannel.html
+++ b/docs/how-to-use-rtcdatachannel.html
@@ -166,7 +166,7 @@
 
 <body>
     <article>
-    <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+    <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
     <h1>How to use RTCDataChannel?
         </h1>
@@ -176,7 +176,7 @@ <h1>How to use RTCDataChannel?
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table style="width: 100%;">
@@ -419,13 +419,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/docs/how-to-use-rtcpeerconnection-js-v1.1.html b/docs/how-to-use-rtcpeerconnection-js-v1.1.html
index 12c4cbf5..d052e56f 100644
--- a/docs/how-to-use-rtcpeerconnection-js-v1.1.html
+++ b/docs/how-to-use-rtcpeerconnection-js-v1.1.html
@@ -166,7 +166,7 @@
 
 <body>
     <article>
-    <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+    <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
     <h1>How to use RTCPeerConnection.js? (v1.1)
         </h1>
@@ -176,7 +176,7 @@ <h1>How to use RTCPeerConnection.js? (v1.1)
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table style="width: 100%;">
@@ -568,7 +568,7 @@ <h2>How to write <a href="#socket-io">same app</a> in WebSocket?
                     </blockquote>
 
                     <pre>
-&lt;script src="<strong>https://webrtc-experiment.appspot.com/dependencies/websocket.js</strong>"&gt;&lt;/script&gt;
+&lt;script src="<strong>https://www.webrtc-experiment.com/dependencies/websocket.js</strong>"&gt;&lt;/script&gt;
 </pre>
                     <br />
                     <blockquote>
@@ -606,13 +606,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/docs/rtc-datachannel-for-beginners.html b/docs/rtc-datachannel-for-beginners.html
index 39854d92..d879bf08 100644
--- a/docs/rtc-datachannel-for-beginners.html
+++ b/docs/rtc-datachannel-for-beginners.html
@@ -166,7 +166,7 @@
 
 <body>
     <article>
-    <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+    <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
     <h1>RTCDataChannel for Beginners
         </h1>
@@ -176,7 +176,7 @@ <h1>RTCDataChannel for Beginners
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table style="width: 100%;">
@@ -624,20 +624,20 @@ <h2>A few tips:</h2>
             <tr>
                 <td>
                     <blockquote>
-                        Are you interested in a "more" simple full-fledged guide? Read <a href="https://webrtc-experiment.appspot.com/docs/how-to-use-rtcpeerconnection-js-v1.1.html">this document</a>.
+                        Are you interested in a "more" simple full-fledged guide? Read <a href="https://www.webrtc-experiment.com/docs/how-to-use-rtcpeerconnection-js-v1.1.html">this document</a>.
                     </blockquote>
                 </td>
             </tr>
             <tr>
                 <td>
                     <blockquote>
-                        Are you interested in a "simple" guide about RTCDataChannel? Read <a href="https://webrtc-experiment.appspot.com/docs/how-to-use-rtcdatachannel.html">this document</a>.
+                        Are you interested in a "simple" guide about RTCDataChannel? Read <a href="https://www.webrtc-experiment.com/docs/how-to-use-rtcdatachannel.html">this document</a>.
                     </blockquote>
                 </td>
             </tr>
 
             <tr>
-                <td>Are you interested to learn <a href="https://webrtc-experiment.appspot.com/docs/how-file-broadcast-works.html" target="_blank">How to share files using RTCDataChannel APIs?</a>
+                <td>Are you interested to learn <a href="https://www.webrtc-experiment.com/docs/how-file-broadcast-works.html" target="_blank">How to share files using RTCDataChannel APIs?</a>
                 </td>
             </tr>
 
@@ -653,13 +653,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/docs/webrtc-for-beginners.html b/docs/webrtc-for-beginners.html
index 49c7fb50..c61403f1 100644
--- a/docs/webrtc-for-beginners.html
+++ b/docs/webrtc-for-beginners.html
@@ -166,7 +166,7 @@
 
 <body>
     <article>
-    <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+    <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
     <h1>WebRTC for Beginners
         </h1>
@@ -176,7 +176,7 @@ <h1>WebRTC for Beginners
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table style="width: 100%;">
@@ -495,13 +495,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/docs/webrtc-for-newbies.html b/docs/webrtc-for-newbies.html
index a26e3065..2e6e78d8 100644
--- a/docs/webrtc-for-newbies.html
+++ b/docs/webrtc-for-newbies.html
@@ -169,7 +169,7 @@
 
 <body>
     <article>
-    <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+    <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
     <h1>WebRTC for newbies!
         </h1>
@@ -179,7 +179,7 @@ <h1>WebRTC for newbies!
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table style="width: 100%;">
@@ -631,13 +631,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/experimental/README.md b/experimental/README.md
index 6ab3c490..1c690e62 100644
--- a/experimental/README.md
+++ b/experimental/README.md
@@ -2,8 +2,8 @@
 
 These experiments are earlier tests of expected updates and latest releases i.e. upcoming features.
 
-----
+=
 
 #### License
 
-Released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+Released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/experimental/mozCaptureStreamUntilEnded/README.md b/experimental/mozCaptureStreamUntilEnded/README.md
index 13a44cad..8aa8203c 100644
--- a/experimental/mozCaptureStreamUntilEnded/README.md
+++ b/experimental/mozCaptureStreamUntilEnded/README.md
@@ -6,14 +6,14 @@ On Firefox, these APIs seems failing to generate correct "ssrc" attributes.
 
 Still no progress.
 
-----
+=
 
 #### Reference
 
 https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pre-recorded-Media-Streaming
 
-----
+=
 
 #### License
 
-Released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+Released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/experimental/mozCaptureStreamUntilEnded/index.html b/experimental/mozCaptureStreamUntilEnded/index.html
index 0f848867..c93b38b4 100644
--- a/experimental/mozCaptureStreamUntilEnded/index.html
+++ b/experimental/mozCaptureStreamUntilEnded/index.html
@@ -158,13 +158,13 @@
     </head>
     
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
 
         <h1>mozCaptureStreamUntilEnded</h1>
 
             <p> <span>Copyright © 2013</span> <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span></p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
 			
 			<video id="pre-recorded-media" src="https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/Pre-recorded-Media-Streaming/WebM-file-to-stream-as-pre-recorded-media.webm" controls autoplay muted style="display:none;"></video>			
@@ -310,13 +310,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                 href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/experimental/remote-media-stream-attachment/README.md b/experimental/remote-media-stream-attachment/README.md
index 75a1c007..88212ba9 100644
--- a/experimental/remote-media-stream-attachment/README.md
+++ b/experimental/remote-media-stream-attachment/README.md
@@ -6,14 +6,14 @@ If it is audio-only streaming; remote audio streams attachment works perfectly f
 
 Still no prgress to correctly forward "video stream".
 
-----
+=
 
 #### Reference
 
 https://github.com/muaz-khan/WebRTC-Experiment/issues/2
 
-----
+=
 
 #### License
 
-Released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+Released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/experimental/remote-media-stream-attachment/index.html b/experimental/remote-media-stream-attachment/index.html
index 1a4b2fbd..07b9c21b 100644
--- a/experimental/remote-media-stream-attachment/index.html
+++ b/experimental/remote-media-stream-attachment/index.html
@@ -158,7 +158,7 @@
     </head>
     
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid red; color: red; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">←HOME</a>
 
         <h1>
 			<a href="https://github.com/muaz-khan/WebRTC-Experiment/blob/master/experimental" target="_blank">Attaching Remote Media Streams</a>
@@ -169,7 +169,7 @@ <h1>
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <div id="videos-container" style="text-align:center;"></div>
 			<script>
@@ -368,13 +368,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                 href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/experimental/remote-stream-recording.html b/experimental/remote-stream-recording.html
index 3c3d2a74..0791d132 100644
--- a/experimental/remote-stream-recording.html
+++ b/experimental/remote-stream-recording.html
@@ -1,4 +1,83 @@
-<script src="/RecordRTC.js"></script>
+<title>Recording Remote Audio Streams / RecordRTC</title>
+<link href='https://fonts.googleapis.com/css?family=Open+Sans' rel='stylesheet' type='text/css'>
+<style>
+html{
+	background: rgb(231, 241, 241);
+	font-family: 'Open Sans', sans-serif;
+    font-size: 1.2em;
+    line-height: 1.5;
+}
+h1 {
+	text-align: center;
+	border-bottom: 2px solid rgb(121, 40, 40);
+	color: rgb(121, 40, 40);
+}
+ol {
+    margin-left: 4em;
+}
+pre {
+    margin-left: 6em;
+}
+</style>
+<h1>Recording Remote Audio Streams / <a href="https://www.webrtc-experiment.com/RecordRTC/">RecordRTC</a></h2>
+
+<h2>issue: <a href="http://stackoverflow.com/questions/17475038/recording-a-remote-webrtc-stream-with-recordrtc/17475416#17475416" target="_blank">unable to record remote audio streams using RecordRTC.</a></h2>
+<blockquote>
+"opus" is the default codec on chrome;
+</blockquote>
+
+<blockquote>
+Chrome uses two modes for opus....
+</blockquote>
+
+<ol>
+<li>"voip" (i.e. voice mode) which is used if mono audio stream is provided by the  sender</li>
+<li>"audio" which is used if stereo audio stream is provided by  the sender</li>
+</ol>
+<blockquote>
+There is no parameter in sdp allows us control voip/audio mode for outgoing/incoming audio streams. This job can be done by sender ( I don't know how, well, still experimenting ).
+</blockquote>
+
+<blockquote>
+They're using code like this:
+</blockquote>
+
+<pre>
+// Default to VoIP application for mono, and AUDIO for stereo.
+int application = (channels == 1) ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO;
+
+state->encoder = opus_encoder_create(48000, channels, application, &error);
+</pre>
+
+<blockquote>
+You can see that if channels equals 1; i.e. if mono audio; they're using "voip" mode. Otherwise "audio" mode.
+</blockquote>
+
+<blockquote>
+In the session description; you can see this line:
+</blockquote>
+
+<pre>
+a=rtpmap:111 opus/48000/2
+</pre>
+
+<blockquote>
+It is mono section with 48000 Hz (i.e. highest) clock rate.
+</blockquote>
+
+<pre>
+Bitrate is 48-64 kb/s .... i.e. FB mono music
+</pre>
+
+<blockquote>
+For stereo; we need:
+</blockquote>
+
+<pre>
+64-128 kb/s .... i.e. FB stereo music
+</pre>
+
+<script src="https://www.webrtc-experiment.com/RecordRTC.js"></script>
 
 <script>
     var offerer, answerer;
@@ -48,31 +127,21 @@
         // stream that is flowing from offerer toward answerer
         answerer.onaddstream = function (event) {
             console.debug('remote stream', event.stream);
-
-            // var audio = document.createElement('audio');
-            // audio.src = URL.createObjectURL(event.stream);
-            // document.body.appendChild(audio);
-            // audio.controls = true;
-            // audio.play();
-			
-			// var context = new webkitAudioContext();
-			// var microphone = context.createMediaStreamSource(event.stream);
-			// var filter = context.createBiquadFilter();
-			// var peer = context.createMediaStreamDestination();
-			// microphone.connect(filter);
-			// filter.connect(peer);
-			
-			// console.log(peer.stream);
 			
-			window.recorder = new RecordRTC({
+			var recorder = new RecordRTC({
 				stream: event.stream
 			});
-			window.recorder.recordAudio();
+			recorder.recordAudio();
 			setTimeout(function() {
-				window.recorder.stopAudio(function() {
-					console.log(window.recorder.getDataURL());
+				recorder.stopAudio(function() {
+					var dataURL = recorder.getDataURL();
+					if(dataURL.indexOf('AAAAAAAAAAAA') != -1) {
+						document.querySelector('h2').innerHTML = 'Unable to record remote audio stream.<br /><br />' +
+																  dataURL.substr(0, 150) + '...';
+					}
+					console.log(dataURL);
 				});
-			}, 15000);
+			}, 5000);
         };
 
         answerer.onicecandidate = function (event) {
@@ -82,8 +151,6 @@
 
         answerer.setRemoteDescription(offer);
         answerer.createAnswer(function (answer) {
-            console.debug('answer sdp', answer.sdp);
-
             answerer.setLocalDescription(answer);
             offerer.setRemoteDescription(answer);
         });
@@ -101,13 +168,4 @@
         }
     }
     getUserMedia(offererPeer);
-	/*
-	getUserMedia(function(stream) {
-		window.recorder = new RecordRTC({
-				stream: stream,
-				audioWorkerPath: 'audio-recorder.js'
-			});
-			window.recorder.recordAudio();
-	});
-	*/
 </script>
\ No newline at end of file
diff --git a/file-hangout/README.md b/file-hangout/README.md
index bc07bfa0..12506738 100644
--- a/file-hangout/README.md
+++ b/file-hangout/README.md
@@ -1,4 +1,4 @@
-#### WebRTC P2P Group File Sharing / [Demo](https://webrtc-experiment.appspot.com/file-hangout/)
+#### WebRTC P2P Group File Sharing / [Demo](https://www.webrtc-experiment.com/file-hangout/)
 
 This WebRTC Experiment allows you share file of any size among group of people.
 
@@ -15,7 +15,7 @@ So, `20` RTP data ports will be opened in `10` users data session. **Embarrassin
 
 On Firefox, by default 16 SCTP data ports will be opened for single peer. So, about 160 SCTP data ports will be opened in 10 users data session. Too awkward!
 
-----
+=
 
 #### Multiple peer connections.....is it a solution?
 
@@ -23,7 +23,7 @@ No, not at all. It is just a **temporary** workaround.
 
 You're strongly suggested to use **peer-to-server** model instead of opening multi-peers.
 
-----
+=
 
 #### How peer-to-server model works?
 
@@ -35,7 +35,7 @@ Remember, WebRTC peer object will send **DTLS/SRTP** packets maybe as **ByteStre
 
 Server can manipulate messages or data coming from 10 or more unique data ports and transfer over single data port!
 
-----
+=
 
 #### ideas!!!
 
@@ -77,11 +77,11 @@ So, we maybe able to send data/text and files over majority of mobile devices.
 
 Though, it seems not possible! Because their infrastructure is different.
 
-----
+=
 
 #### Browser Support
 
-WebRTC [Group File Sharing](https://webrtc-experiment.appspot.com/file-hangout/) experiment works fine on following web-browsers:
+WebRTC [Group File Sharing](https://www.webrtc-experiment.com/file-hangout/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -89,8 +89,8 @@ WebRTC [Group File Sharing](https://webrtc-experiment.appspot.com/file-hangout/)
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-WebRTC [Group File Sharing](https://webrtc-experiment.appspot.com/file-hangout/) experiment is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+WebRTC [Group File Sharing](https://www.webrtc-experiment.com/file-hangout/) experiment is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/file-hangout/hangout-ui.js b/file-hangout/hangout-ui.js
index bfda92c1..82ca10f7 100644
--- a/file-hangout/hangout-ui.js
+++ b/file-hangout/hangout-ui.js
@@ -1,25 +1,36 @@
-/*
- 2013, @muazkh » github.com/muaz-khan
- MIT License » https://webrtc-experiment.appspot.com/licence/
- Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel
-*/
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-hangout
 
 var config = {
-    openSocket: function (config) {
-        var channel = config.channel || location.hash.replace('#', '') || 'file-hangout';
-        var socket = new Firebase('https://rtcweb.firebaseIO.com/' + channel);
+    openSocket: function(config) {
+        var SIGNALING_SERVER = 'https://www.webrtc-experiment.com:8553/',
+            defaultChannel = location.hash.substr(1) || 'group-file-sharing-hangout';
+
+        var channel = config.channel || defaultChannel;
+        var sender = Math.round(Math.random() * 999999999) + 999999999;
+
+        io.connect(SIGNALING_SERVER).emit('new-channel', {
+            channel: channel,
+            sender: sender
+        });
+
+        var socket = io.connect(SIGNALING_SERVER + channel);
         socket.channel = channel;
-        socket.on('child_added', function (data) {
-            config.onmessage(data.val());
+        socket.on('connect', function() {
+            if (config.callback) config.callback(socket);
         });
-        socket.send = function (data) {
-            this.push(data);
-        }
-        config.onopen && setTimeout(config.onopen, 1);
-        socket.onDisconnect().remove();
-        return socket;
+
+        socket.send = function(message) {
+            socket.emit('message', {
+                sender: sender,
+                data: message
+            });
+        };
+
+        socket.on('message', config.onmessage);
     },
-    onRoomFound: function (room) {
+    onRoomFound: function(room) {
         var alreadyExist = document.getElementById(room.broadcaster);
         if (alreadyExist) return;
 
@@ -33,7 +44,7 @@ var config = {
 
         roomsList.insertBefore(tr, roomsList.firstChild);
 
-        tr.onclick = function () {
+        tr.onclick = function() {
             var tr = this;
             hangoutUI.joinRoom({
                 roomToken: tr.querySelector('.join').id,
@@ -43,24 +54,23 @@ var config = {
             hideUnnecessaryStuff();
         };
     },
-    onChannelOpened: function (/* channel */) {
+    onChannelOpened: function(/* channel */) {
         unnecessaryStuffVisible && hideUnnecessaryStuff();
         if (fileElement) fileElement.removeAttribute('disabled');
     },
-    onChannelMessage: function (data) {
+    onChannelMessage: function(data) {
         if (data.sender && participants) {
             var tr = document.createElement('tr');
             tr.innerHTML = '<td>' + data.sender + ' is ready to receive files!</td>';
             participants.insertBefore(tr, participants.firstChild);
-        }
-        else onMessageCallback(data);
+        } else onMessageCallback(data);
     }
 };
 
 function createButtonClickHandler() {
     hangoutUI.createRoom({
         userName: prompt('Enter your name', 'Anonymous'),
-        roomName: ((document.getElementById('conference-name') || {}).value || 'Anonymous') + ' // shared via ' + (navigator.vendor ? 'Google Chrome (Stable/Canary)' : 'Mozilla Firefox (Aurora/Nightly)')
+        roomName: ((document.getElementById('conference-name') || { }).value || 'Anonymous') + ' // shared via ' + (!!navigator.webkitGetUserMedia ? 'Google Chrome (Stable/Canary)' : 'Mozilla Firefox (Aurora/Nightly)')
     });
     hideUnnecessaryStuff();
 }
@@ -78,6 +88,7 @@ var roomsList = document.getElementById('rooms-list');
 var chatOutput = document.getElementById('chat-output');
 
 var unnecessaryStuffVisible = true;
+
 function hideUnnecessaryStuff() {
     var visibleElements = document.getElementsByClassName('visible'),
         length = visibleElements.length;
@@ -91,7 +102,7 @@ function hideUnnecessaryStuff() {
 
 var chatMessage = document.getElementById('chat-message');
 if (chatMessage)
-    chatMessage.onchange = function () {
+    chatMessage.onchange = function() {
         hangoutUI.send(chatMessage.value);
         chatMessage.value = '';
     };
@@ -111,7 +122,7 @@ function onMessageCallback(data) {
     if (data.size && moz) {
         var reader = new window.FileReader();
         reader.readAsDataURL(data);
-        reader.onload = function (event) {
+        reader.onload = function(event) {
             saveToDisk(event.target.result, lastFileName);
             quickOutput(lastFileName, 'received successfully!');
             disable(false);
@@ -149,7 +160,7 @@ function onMessageCallback(data) {
 
 // getting file from user's system
 var file, fileElement = document.getElementById('file');
-fileElement.onchange = function () {
+fileElement.onchange = function() {
     file = fileElement.files[0];
     if (!file) return false;
 
@@ -157,7 +168,7 @@ fileElement.onchange = function () {
     if (moz) {
         hangoutUI.send(JSON.stringify({ lastFileName: file.name }));
         quickOutput(file.name, 'shared successfully!');
-        setTimeout(function () {
+        setTimeout(function() {
             if (fileElement) fileElement.value = '';
         }, 0);
         return hangoutUI.send(file);
@@ -170,8 +181,9 @@ fileElement.onchange = function () {
 };
 
 var packetSize = 1000, textToTransfer = '', packets = 0;
+
 function onReadAsDataURL(evt, text) {
-    var data = {};
+    var data = { };
 
     if (evt) {
         text = evt.target.result;
@@ -190,7 +202,7 @@ function onReadAsDataURL(evt, text) {
         quickOutput(file.name, 'shared successfully!');
 
         disable(false);
-        setTimeout(function () {
+        setTimeout(function() {
             if (fileElement) fileElement.value = '';
         }, 0);
     }
@@ -199,7 +211,7 @@ function onReadAsDataURL(evt, text) {
     textToTransfer = text.slice(data.message.length);
 
     if (textToTransfer.length)
-        setTimeout(function () {
+        setTimeout(function() {
             onReadAsDataURL(null, textToTransfer);
         }, 500);
 }
@@ -220,6 +232,7 @@ function saveToDisk(fileUrl, fileName) {
 
 // UI
 var outputPanel = document.getElementById('output-panel');
+
 function quickOutput(message, message2) {
     if (!outputPanel) return;
     if (message2) message = '<strong>' + message + '</strong> ' + message2;
@@ -230,20 +243,22 @@ function quickOutput(message, message2) {
 }
 
 var statusDiv = document.getElementById('status');
+
 function updateStatus() {
     packets--;
     if (statusDiv) statusDiv.innerHTML = packets + ' items remaining.';
     if (packets <= 0) statusDiv.innerHTML = '';
 }
 
-(function () {
+(function() {
     var uniqueToken = document.getElementById('unique-token');
-    if (uniqueToken) if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
-    else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace(/\./g, '-');
+    if (uniqueToken)
+        if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
+        else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace( /\./g , '-');
 })();
 
 function disable(_disable) {
     if (!fileElement) return;
     if (!_disable) fileElement.removeAttribute('disabled');
     else fileElement.setAttribute('disabled', true);
-}
\ No newline at end of file
+}
diff --git a/file-hangout/hangout.js b/file-hangout/hangout.js
index a94f7a10..50d4398c 100644
--- a/file-hangout/hangout.js
+++ b/file-hangout/hangout.js
@@ -1,6 +1,8 @@
-/* MIT License: https://webrtc-experiment.appspot.com/licence/ */
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-hangout
 
-var hangout = function (config) {
+var hangout = function(config) {
     var self = {
         userToken: uniqueToken(),
         userName: 'Anonymous'
@@ -10,222 +12,210 @@ var hangout = function (config) {
         sockets = [],
         isGetNewRoom = true;
 
-    var defaultSocket = {}, RTCDataChannels = []
+    var defaultSocket = { }, RTCDataChannels = [];
 
-        function openDefaultSocket() {
-            defaultSocket = config.openSocket({
-                    onmessage: onDefaultSocketResponse,
-                    callback: function (socket) {
-                        defaultSocket = socket;
-                    }
-                });
-        }
+    function openDefaultSocket() {
+        defaultSocket = config.openSocket({
+            onmessage: onDefaultSocketResponse,
+            callback: function(socket) {
+                defaultSocket = socket;
+            }
+        });
+    }
 
-        function onDefaultSocketResponse(response) {
-            if (response.userToken == self.userToken) return;
+    function onDefaultSocketResponse(response) {
+        if (response.userToken == self.userToken) return;
 
-            if (isGetNewRoom && response.roomToken && response.broadcaster) config.onRoomFound(response);
+        if (isGetNewRoom && response.roomToken && response.broadcaster) config.onRoomFound(response);
 
-            if (response.newParticipant) onNewParticipant(response.newParticipant);
+        if (response.newParticipant) onNewParticipant(response.newParticipant);
 
-            if (response.userToken && response.joinUser == self.userToken && response.participant && channels.indexOf(response.userToken) == -1) {
-                channels += response.userToken + '--';
-                openSubSocket({
-                        isofferer: true,
-                        channel: response.channel || response.userToken,
-                        closeSocket: true
-                    });
-            }
+        if (response.userToken && response.joinUser == self.userToken && response.participant && channels.indexOf(response.userToken) == -1) {
+            channels += response.userToken + '--';
+            openSubSocket({
+                isofferer: true,
+                channel: response.channel || response.userToken,
+                closeSocket: true
+            });
         }
+    }
 
-        function openSubSocket(_config) {
-            if (!_config.channel) return;
-            var socketConfig = {
-                channel: _config.channel,
-                onmessage: socketResponse,
-                onopen: function () {
-                    if (isofferer && !peer) initPeer();
-                    sockets[sockets.length] = socket;
-                }
-            };
-
-            socketConfig.callback = function (_socket) {
-                socket = _socket;
-                this.onopen();
-            };
-
-            var socket = config.openSocket(socketConfig),
-                isofferer = _config.isofferer,
-                gotstream,
-                inner = {},
-                peer;
-
-            var peerConfig = {
-                onICE: function (candidate) {
-                    socket.send({
-                            userToken: self.userToken,
-                            candidate: {
-                                sdpMLineIndex: candidate.sdpMLineIndex,
-                                candidate: JSON.stringify(candidate.candidate)
-                            }
-                        });
-                },
-                onChannelOpened: onChannelOpened,
-                onChannelMessage: function (event) {
-                    config.onChannelMessage(event.data.size ? event.data : JSON.parse(event.data));
-                }
-            };
-
-            function initPeer(offerSDP) {
-                if (!offerSDP) {
-                    peerConfig.onOfferSDP = sendsdp;
-                } else {
-                    peerConfig.offerSDP = offerSDP;
-                    peerConfig.onAnswerSDP = sendsdp;
-                }
+    function openSubSocket(_config) {
+        if (!_config.channel) return;
+        var socketConfig = {
+            channel: _config.channel,
+            onmessage: socketResponse,
+            onopen: function() {
+                if (isofferer && !peer) initPeer();
+                sockets[sockets.length] = socket;
+            }
+        };
+
+        socketConfig.callback = function(_socket) {
+            socket = _socket;
+            this.onopen();
+        };
 
-                peer = RTCPeerConnection(peerConfig);
+        var socket = config.openSocket(socketConfig),
+            isofferer = _config.isofferer,
+            gotstream,
+            inner = { },
+            peer;
+
+        var peerConfig = {
+            onICE: function(candidate) {
+                socket.send({
+                    userToken: self.userToken,
+                    candidate: {
+                        sdpMLineIndex: candidate.sdpMLineIndex,
+                        candidate: JSON.stringify(candidate.candidate)
+                    }
+                });
+            },
+            onChannelOpened: onChannelOpened,
+            onChannelMessage: function(event) {
+                config.onChannelMessage(event.data.size ? event.data : JSON.parse(event.data));
             }
+        };
 
-            function onChannelOpened(channel) {
-                RTCDataChannels[RTCDataChannels.length] = channel;
-                channel.send(JSON.stringify({
-                            sender: self.userName
-                        }));
+        function initPeer(offerSDP) {
+            if (!offerSDP) {
+                peerConfig.onOfferSDP = sendsdp;
+            } else {
+                peerConfig.offerSDP = offerSDP;
+                peerConfig.onAnswerSDP = sendsdp;
+            }
 
-                if (config.onChannelOpened) config.onChannelOpened(channel);
+            peer = RTCPeerConnection(peerConfig);
+        }
 
-                if (isbroadcaster && channels.split('--').length > 3) {
-                    /* broadcasting newly connected participant for video-conferencing! */
-                    defaultSocket.send({
-                            newParticipant: socket.channel,
-                            userToken: self.userToken
-                        });
-                }
+        function onChannelOpened(channel) {
+            RTCDataChannels[RTCDataChannels.length] = channel;
+            channel.send(JSON.stringify({
+                sender: self.userName
+            }));
 
-                /* closing subsocket here on the offerer side */
-                if (_config.closeSocket) socket = null;
+            if (config.onChannelOpened) config.onChannelOpened(channel);
 
-                gotstream = true;
+            if (isbroadcaster && channels.split('--').length > 3) {
+                /* broadcasting newly connected participant for video-conferencing! */
+                defaultSocket.send({
+                    newParticipant: socket.channel,
+                    userToken: self.userToken
+                });
             }
 
-            function sendsdp(sdp) {
-                sdp = JSON.stringify(sdp);
-                var part = parseInt(sdp.length / 3);
+            /* closing subsocket here on the offerer side */
+            if (_config.closeSocket) socket = null;
 
-                var firstPart = sdp.slice(0, part),
-                    secondPart = sdp.slice(part, sdp.length - 1),
-                    thirdPart = '';
-
-                if (sdp.length > part + part) {
-                    secondPart = sdp.slice(part, part + part);
-                    thirdPart = sdp.slice(part + part, sdp.length);
-                }
+            gotstream = true;
+        }
 
-                socket.send({
-                        userToken: self.userToken,
-                        firstPart: firstPart
-                    });
+        function sendsdp(sdp) {
+            sdp = JSON.stringify(sdp);
+            var part = parseInt(sdp.length / 3);
 
-                socket.send({
-                        userToken: self.userToken,
-                        secondPart: secondPart
-                    });
+            var firstPart = sdp.slice(0, part),
+                secondPart = sdp.slice(part, sdp.length - 1),
+                thirdPart = '';
 
-                socket.send({
-                        userToken: self.userToken,
-                        thirdPart: thirdPart
-                    });
+            if (sdp.length > part + part) {
+                secondPart = sdp.slice(part, part + part);
+                thirdPart = sdp.slice(part + part, sdp.length);
             }
 
-            function socketResponse(response) {
-                if (response.userToken == self.userToken) return;
+            socket.send({
+                userToken: self.userToken,
+                firstPart: firstPart
+            });
 
-                if (response.firstPart || response.secondPart || response.thirdPart) {
-                    if (response.firstPart) {
-                        inner.firstPart = response.firstPart;
-                        if (inner.secondPart && inner.thirdPart) selfInvoker();
-                    }
-                    if (response.secondPart) {
-                        inner.secondPart = response.secondPart;
-                        if (inner.firstPart && inner.thirdPart) selfInvoker();
-                    }
+            socket.send({
+                userToken: self.userToken,
+                secondPart: secondPart
+            });
 
-                    if (response.thirdPart) {
-                        inner.thirdPart = response.thirdPart;
-                        if (inner.firstPart && inner.secondPart) selfInvoker();
-                    }
+            socket.send({
+                userToken: self.userToken,
+                thirdPart: thirdPart
+            });
+        }
+
+        function socketResponse(response) {
+            if (response.userToken == self.userToken) return;
+
+            if (response.firstPart || response.secondPart || response.thirdPart) {
+                if (response.firstPart) {
+                    inner.firstPart = response.firstPart;
+                    if (inner.secondPart && inner.thirdPart) selfInvoker();
+                }
+                if (response.secondPart) {
+                    inner.secondPart = response.secondPart;
+                    if (inner.firstPart && inner.thirdPart) selfInvoker();
                 }
 
-                if (response.candidate && !gotstream) {
-                    peer && peer.addICE({
-                            sdpMLineIndex: response.candidate.sdpMLineIndex,
-                            candidate: JSON.parse(response.candidate.candidate)
-                        });
+                if (response.thirdPart) {
+                    inner.thirdPart = response.thirdPart;
+                    if (inner.firstPart && inner.secondPart) selfInvoker();
                 }
+            }
 
-                if (response.left) {
-                    if (peer && peer.peer) {
-                        peer.peer.close();
-                        peer.peer = null;
-                    }
+            if (response.candidate && !gotstream) {
+                peer && peer.addICE({
+                    sdpMLineIndex: response.candidate.sdpMLineIndex,
+                    candidate: JSON.parse(response.candidate.candidate)
+                });
+            }
+
+            if (response.left) {
+                if (peer && peer.peer) {
+                    peer.peer.close();
+                    peer.peer = null;
                 }
             }
+        }
 
-            var invokedOnce = false;
+        var invokedOnce = false;
 
-            function selfInvoker() {
-                if (invokedOnce) return;
+        function selfInvoker() {
+            if (invokedOnce) return;
 
-                invokedOnce = true;
+            invokedOnce = true;
 
-                inner.sdp = JSON.parse(inner.firstPart + inner.secondPart + inner.thirdPart);
+            inner.sdp = JSON.parse(inner.firstPart + inner.secondPart + inner.thirdPart);
 
-                if (isofferer) peer.addAnswerSDP(inner.sdp);
-                else initPeer(inner.sdp);
-            }
+            if (isofferer) peer.addAnswerSDP(inner.sdp);
+            else initPeer(inner.sdp);
         }
+    }
 
-        function leave() {
-            length = sockets.length;
-            for (var i = 0; i < length; i++) {
-                socket = sockets[i];
-                if (socket) {
-                    socket.send({
-                            left: true,
-                            userToken: self.userToken
-                        });
-                    delete sockets[i];
-                }
+    function leave() {
+        var length = sockets.length;
+        for (var i = 0; i < length; i++) {
+            var socket = sockets[i];
+            if (socket) {
+                socket.send({
+                    left: true,
+                    userToken: self.userToken
+                });
+                delete sockets[i];
             }
         }
+    }
 
-    window.onunload = function () {
+    window.onunload = function() {
         leave();
     };
 
-    window.onkeyup = function (e) {
+    window.onkeyup = function(e) {
         if (e.keyCode == 116) leave();
     };
 
-    (function () {
-            var anchors = document.querySelectorAll('a'),
-                length = anchors.length;
-            for (var i = 0; i < length; i++) {
-                a = anchors[i];
-                if (a.href.indexOf('#') !== 0 && a.getAttribute('target') != '_blank')
-                    a.onclick = function () {
-                        leave();
-                };
-            }
-        })();
-
     function startBroadcasting() {
         defaultSocket && defaultSocket.send({
-                roomToken: self.roomToken,
-                roomName: self.roomName,
-                broadcaster: self.userToken
-            });
+            roomToken: self.roomToken,
+            roomName: self.roomName,
+            broadcaster: self.userToken
+        });
         setTimeout(startBroadcasting, 3000);
     }
 
@@ -235,20 +225,20 @@ var hangout = function (config) {
 
         var new_channel = uniqueToken();
         openSubSocket({
-                channel: new_channel,
-                closeSocket: true
-            });
+            channel: new_channel,
+            closeSocket: true
+        });
 
         defaultSocket.send({
-                participant: true,
-                userToken: self.userToken,
-                joinUser: channel,
-                channel: new_channel
-            });
+            participant: true,
+            userToken: self.userToken,
+            joinUser: channel,
+            channel: new_channel
+        });
     }
 
     function uniqueToken() {
-        var s4 = function () {
+        var s4 = function() {
             return Math.floor(Math.random() * 0x10000).toString(16);
         };
         return s4() + s4() + "-" + s4() + "-" + s4() + "-" + s4() + "-" + s4() + s4() + s4();
@@ -256,7 +246,7 @@ var hangout = function (config) {
 
     openDefaultSocket();
     return {
-        createRoom: function (_config) {
+        createRoom: function(_config) {
             self.roomName = _config.roomName || 'Anonymous';
             self.roomToken = uniqueToken();
             if (_config.userName) self.userName = _config.userName;
@@ -265,26 +255,29 @@ var hangout = function (config) {
             isGetNewRoom = false;
             startBroadcasting();
         },
-        joinRoom: function (_config) {
+        joinRoom: function(_config) {
             self.roomToken = _config.roomToken;
             if (_config.userName) self.userName = _config.userName;
             isGetNewRoom = false;
 
             openSubSocket({
-                    channel: self.userToken
-                });
+                channel: self.userToken
+            });
 
             defaultSocket.send({
-                    participant: true,
-                    userToken: self.userToken,
-                    joinUser: _config.joinUser
-                });
+                participant: true,
+                userToken: self.userToken,
+                joinUser: _config.joinUser
+            });
         },
-        send: function (data) {
+        send: function(data) {
             var length = RTCDataChannels.length;
             if (!length) return;
             for (var i = 0; i < length; i++) {
-                RTCDataChannels[i].send(data);
+                try {
+                    RTCDataChannels[i].send(data);
+                } catch(e) {
+                }
             }
         }
     };
diff --git a/file-hangout/index.html b/file-hangout/index.html
index 5c77e12a..0d18346a 100644
--- a/file-hangout/index.html
+++ b/file-hangout/index.html
@@ -178,7 +178,7 @@
 
 <body>
 <article>
-    <a href="https://webrtc-experiment.appspot.com/"
+    <a href="https://www.webrtc-experiment.com/"
        style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
         WEBRTC EXPERIMENTS</a>
 
@@ -191,7 +191,7 @@ <h1><a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-han
     </p>
 
     <section class="plusone-gplus">
-        <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+        <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
     </section>
 
     <table style="width: 100%;">
@@ -219,10 +219,10 @@ <h1><a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/file-han
         </tr>
     </table>
 
-    <script src='https://webrtc-experiment.appspot.com/firebase.js'></script>
-    <script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/file-hangout/hangout.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/file-hangout/hangout-ui.js"></script>
+    <script src="https://www.webrtc-experiment.com/socket.io.js"></script>
+    <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
+    <script src="https://www.webrtc-experiment.com/file-hangout/hangout.js"></script>
+    <script src="https://www.webrtc-experiment.com/file-hangout/hangout-ui.js"></script>
 
     <br/>
     <br/>
@@ -240,7 +240,7 @@ <h2>How it works?</h2>
         <li>in reality, it is peer-to-peer-to-peers....</li>
     </ol>
 
-    <p>Interested to learn <a href="https://webrtc-experiment.appspot.com/docs/how-file-broadcast-works.html"
+    <p>Interested to learn <a href="https://www.webrtc-experiment.com/docs/how-file-broadcast-works.html"
                                target="_blank">how to share files using RTCDataChannel APIs?</a></p>
  </article>
  <article>
@@ -259,7 +259,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
 </article>
 <footer>
     <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+        <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
         <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
         <span>2013 </span>»
         <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -267,6 +267,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         <a href="https://github.com/muaz-khan" target="_blank">Github</a>
     </p>
 </footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+<script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/file-sharing/README.md b/file-sharing/README.md
index ac0bbbe5..c161ebdc 100644
--- a/file-sharing/README.md
+++ b/file-sharing/README.md
@@ -1,4 +1,4 @@
-#### WebRTC File Sharing i.e. Data Sharing / [Demo](https://webrtc-experiment.appspot.com/file-sharing/)
+#### WebRTC File Sharing i.e. Data Sharing / [Demo](https://www.webrtc-experiment.com/file-sharing/)
 
 `DataConnection.js` library lets you:
 
@@ -6,15 +6,15 @@
 2. Share text message of any length
 3. Text data regardless of the size and type
 
-----
+=
 
 #### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/data-connection.js"></script>
+<script src="https://www.webrtc-experiment.com/data-connection.js"></script>
 ```
 
-----
+=
 
 #### Last Step: Start using it!
 
@@ -29,7 +29,7 @@ document.getElementById('setup-new-connection').onclick = function() {
 };
 ```
 
-----
+=
 
 #### Text Chat i.e. Text Sharing
 
@@ -43,7 +43,7 @@ You may want to share direct messages:
 connection.channels['user-id'].send('longest possible text message');
 ```
 
-----
+=
 
 #### File Sharing
 
@@ -84,7 +84,7 @@ channel.onFileSent = function (e) {
 channel.onFileReceived = function (e) { /* e.fileName, e.userid */ };
 ```
 
-----
+=
 
 #### Errors Handling
 
@@ -96,7 +96,7 @@ connection.onerror = function(e) {}
 connection.onclose = function(e) {}
 ```
 
-----
+=
 
 #### Custom user-ids?
 
@@ -104,7 +104,7 @@ connection.onclose = function(e) {}
 connection.userid = 'username';
 ```
 
-----
+=
 
 #### Custom signaling channel?
 
@@ -143,7 +143,7 @@ connection.firebase = 'chat';
 
 Want to use XHR, WebSockets, SIP, XMPP, etc. for signaling? Read [this post](https://github.com/muaz-khan/WebRTC-Experiment/issues/56#issuecomment-20090650).
 
-----
+=
 
 #### Want to manually join rooms?
 
@@ -158,7 +158,7 @@ connection.onconnection = function(room) {
 
 `onconnection` is called for each new data connection; and `join` method allows you manually join previously created connections.
 
-----
+=
 
 #### If someone leaves...
 
@@ -170,11 +170,11 @@ connection.onuserleft = function(userid) {
 };
 ```
 
-----
+=
 
 #### Browser Support
 
-This [DataConnection.js](https://webrtc-experiment.appspot.com/data-connection.js) library is compatible to following web-browsers:
+This [DataConnection.js](https://www.webrtc-experiment.com/data-connection.js) library is compatible to following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -182,8 +182,8 @@ This [DataConnection.js](https://webrtc-experiment.appspot.com/data-connection.j
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[DataConnection.js](https://webrtc-experiment.appspot.com/data-connection.js) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[DataConnection.js](https://www.webrtc-experiment.com/data-connection.js) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/file-sharing/index.html b/file-sharing/index.html
index 4e5749f9..35e6ae7c 100644
--- a/file-sharing/index.html
+++ b/file-sharing/index.html
@@ -132,8 +132,8 @@
                 margin-left: 6em;
             }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/data-connection.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/data-connection.js"> </script>
     </head>
     <body>
         <h1>
@@ -145,7 +145,7 @@ <h1>
                                               Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <section>
             <h2>Setup a new connection:</h2>
@@ -246,7 +246,7 @@ <h2 style="display: block; font-size: 1em; text-align: center;">
         <br />
         <br />
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/data-connection.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/data-connection.js"&gt;&lt;/script&gt;
 </pre>
         <pre>
 var connection = new DataConnection();
@@ -283,8 +283,8 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
diff --git a/meeting/README.md b/meeting/README.md
index 0d758600..01b121ae 100644
--- a/meeting/README.md
+++ b/meeting/README.md
@@ -1,4 +1,4 @@
-#### WebRTC Meeting i.e. Video-Conferencing / [Demo](https://webrtc-experiment.appspot.com/meeting/)
+#### WebRTC Meeting i.e. Video-Conferencing / [Demo](https://www.webrtc-experiment.com/meeting/)
 
 1. Multiple peer-connections are opened to bring multi-users connectivity experience.
 2. Maximum peers limit on chrome is temporarily 10.
@@ -21,15 +21,15 @@ Possible issues:
 
 Solution? Obviously a media server. To overcome burden and to deliver HD stream over thousands of peers; we need a media server that should broadcast stream over number of peers.
 
-----
+=
 
 #### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/meeting/meeting.js"></script>
+<script src="https://www.webrtc-experiment.com/meeting/meeting.js"></script>
 ```
 
-----
+=
 
 #### Last Step: Start using it!
 
@@ -53,7 +53,7 @@ document.getElementById('setup-new-meeting').onclick = function() {
 };
 ```
 
-----
+=
 
 #### Custom user-ids?
 
@@ -61,7 +61,7 @@ document.getElementById('setup-new-meeting').onclick = function() {
 meeting.userid = 'username';
 ```
 
-----
+=
 
 #### Custom signaling channel?
 
@@ -98,7 +98,7 @@ Want to use `Firebase` for signaling?
 meeting.firebase = 'chat';
 ```
 
-----
+=
 
 #### Want to manually join rooms?
 
@@ -119,7 +119,7 @@ meeting.onmeeting = function(room) {
 
 `onmeeting` is called for each new meeting; and `meet` method allows you manually join a meeting room.
 
-----
+=
 
 #### If someone leaves...
 
@@ -133,7 +133,7 @@ meeting.onuserleft = function(userid) {
 };
 ```
 
-----
+=
 
 #### `onaddstream`
 
@@ -152,11 +152,11 @@ meeting.onaddstream = function(e) {
 };
 ```
 
-----
+=
 
 #### Browser Support
 
-This [WebRTC Meeting](https://webrtc-experiment.appspot.com/meeting/) experiment works fine on following web-browsers:
+This [WebRTC Meeting](https://www.webrtc-experiment.com/meeting/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -164,8 +164,8 @@ This [WebRTC Meeting](https://webrtc-experiment.appspot.com/meeting/) experiment
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[WebRTC Meeting](https://webrtc-experiment.appspot.com/meeting/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC Meeting](https://www.webrtc-experiment.com/meeting/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/meeting/index.html b/meeting/index.html
index 03e21755..aae6b553 100644
--- a/meeting/index.html
+++ b/meeting/index.html
@@ -108,8 +108,8 @@
                 padding-left: 1em;
             }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/meeting/meeting.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/meeting/meeting.js"> </script>
     </head>
     <body>
         <h1>
@@ -121,7 +121,7 @@ <h1>
                                               Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <section>
             <h2>Setup a new meeting:</h2>
@@ -211,7 +211,7 @@ <h2 style="display: block; text-align: center;">
         <br />
         <br />
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/meeting/meeting.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/meeting/meeting.js"&gt;&lt;/script&gt;
 </pre>
         <pre>
 var meeting = new Meeting('meeting-unique-id');
@@ -258,8 +258,8 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
\ No newline at end of file
diff --git a/one-to-many-audio-broadcasting/README.md b/one-to-many-audio-broadcasting/README.md
index 8105abaf..07b534a2 100644
--- a/one-to-many-audio-broadcasting/README.md
+++ b/one-to-many-audio-broadcasting/README.md
@@ -1,11 +1,11 @@
-#### WebRTC One-to-Many audio-broadcasting / [Demo](https://webrtc-experiment.appspot.com/one-to-many-audio-broadcasting/)
+#### WebRTC One-to-Many audio-broadcasting / [Demo](https://www.webrtc-experiment.com/one-to-many-audio-broadcasting/)
 
 If 10 users join your broadcasted room, **20 RTP ports** will be opened on your browser:
 
 1. 10 RTP ports for **outgoing** audio streams
 2. 10 RTP ports for **incoming** audio streams
 
-----
+=
 
 #### Difference between one-way broadcasting and one-to-many broadcasting
 
@@ -15,15 +15,15 @@ On each participant's side; only one **incoming** RTP port will be opened.
 
 Unlike one-way broadcasting; one-to-many broadcasting experiment opens both outgoing as well as incoming RTP ports for each participant.
 
-----
+=
 
 #### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/one-to-many-audio-broadcasting/meeting.js"></script>
+<script src="https://www.webrtc-experiment.com/one-to-many-audio-broadcasting/meeting.js"></script>
 ```
 
-----
+=
 
 #### Last Step: Start using it!
 
@@ -47,7 +47,7 @@ document.getElementById('setup-new-meeting').onclick = function() {
 };
 ```
 
-----
+=
 
 #### Custom user-ids?
 
@@ -55,7 +55,7 @@ document.getElementById('setup-new-meeting').onclick = function() {
 meeting.userid = 'username';
 ```
 
-----
+=
 
 #### Custom signaling channel?
 
@@ -92,7 +92,7 @@ Want to use `Firebase` for signaling?
 meeting.firebase = 'chat';
 ```
 
-----
+=
 
 #### Want to manually join rooms?
 
@@ -113,7 +113,7 @@ meeting.onmeeting = function(room) {
 
 `onmeeting` is called for each new meeting; and `meet` method allows you manually join a meeting room.
 
-----
+=
 
 #### If someone leaves...
 
@@ -127,8 +127,7 @@ meeting.onuserleft = function(userid) {
 };
 ```
 
-----
-
+=
 #### `onaddstream`
 
 It is called both for `local` and `remote` media streams. It returns:
@@ -146,11 +145,11 @@ meeting.onaddstream = function(e) {
 };
 ```
 
-----
+=
 
 #### Browser Support
 
-This [WebRTC One-to-Many audio-broadcasting](https://webrtc-experiment.appspot.com/one-to-many-audio-broadcasting/) experiment works fine on following web-browsers:
+This [WebRTC One-to-Many audio-broadcasting](https://www.webrtc-experiment.com/one-to-many-audio-broadcasting/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -158,8 +157,8 @@ This [WebRTC One-to-Many audio-broadcasting](https://webrtc-experiment.appspot.c
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[WebRTC One-to-Many audio-broadcasting](https://webrtc-experiment.appspot.com/one-to-many-audio-broadcasting/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC One-to-Many audio-broadcasting](https://www.webrtc-experiment.com/one-to-many-audio-broadcasting/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/one-to-many-audio-broadcasting/index.html b/one-to-many-audio-broadcasting/index.html
index d0466679..c862c160 100644
--- a/one-to-many-audio-broadcasting/index.html
+++ b/one-to-many-audio-broadcasting/index.html
@@ -108,8 +108,8 @@
                 padding-left: 1em;
             }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/one-to-many-audio-broadcasting/meeting.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/one-to-many-audio-broadcasting/meeting.js"> </script>
     </head>
     <body>
         <h1>
@@ -120,7 +120,7 @@ <h1>
                                               Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <section>
             <h2>Setup a new meeting:</h2>
@@ -192,7 +192,7 @@ <h2>Difference between one-way broadcasting and one-to-many broadcasting</h2>
 		<br />
         <br />
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/one-to-many-audio-broadcasting/meeting.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/one-to-many-audio-broadcasting/meeting.js"&gt;&lt;/script&gt;
 </pre>
         <pre>
 var meeting = new Meeting('meeting-unique-id');
@@ -239,8 +239,8 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
\ No newline at end of file
diff --git a/one-to-many-video-broadcasting/README.md b/one-to-many-video-broadcasting/README.md
index c7444c73..4e88f27e 100644
--- a/one-to-many-video-broadcasting/README.md
+++ b/one-to-many-video-broadcasting/README.md
@@ -1,4 +1,4 @@
-#### WebRTC One-to-Many video-broadcasting / [Demo](https://webrtc-experiment.appspot.com/one-to-many-video-broadcasting/)
+#### WebRTC One-to-Many video-broadcasting / [Demo](https://www.webrtc-experiment.com/one-to-many-video-broadcasting/)
 
 If 10 users join your broadcasted room, **40 RTP ports** will be opened on your browser:
 
@@ -7,7 +7,7 @@ If 10 users join your broadcasted room, **40 RTP ports** will be opened on your
 3. 10 RTP ports for **incoming** audio streams
 4. 10 RTP ports for **incoming** video streams
 
-----
+=
 
 #### Difference between one-way broadcasting and one-to-many broadcasting
 
@@ -22,15 +22,15 @@ On each participant's side; only 2 **incoming** RTP ports will be opened.
 
 Unlike one-way broadcasting; one-to-many broadcasting experiment opens both outgoing as well as incoming RTP ports for each participant.
 
-----
+=
 
 #### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/one-to-many-video-broadcasting/meeting.js"></script>
+<script src="https://www.webrtc-experiment.com/one-to-many-video-broadcasting/meeting.js"></script>
 ```
 
-----
+=
 
 #### Last Step: Start using it!
 
@@ -54,7 +54,7 @@ document.getElementById('setup-new-meeting').onclick = function() {
 };
 ```
 
-----
+=
 
 #### Custom user-ids?
 
@@ -62,7 +62,7 @@ document.getElementById('setup-new-meeting').onclick = function() {
 meeting.userid = 'username';
 ```
 
-----
+=
 
 #### Custom signaling channel?
 
@@ -99,7 +99,7 @@ Want to use `Firebase` for signaling?
 meeting.firebase = 'chat';
 ```
 
-----
+=
 
 #### Want to manually join rooms?
 
@@ -120,7 +120,7 @@ meeting.onmeeting = function(room) {
 
 `onmeeting` is called for each new meeting; and `meet` method allows you manually join a meeting room.
 
-----
+=
 
 #### If someone leaves...
 
@@ -134,7 +134,7 @@ meeting.onuserleft = function(userid) {
 };
 ```
 
-----
+=
 
 #### `onaddstream`
 
@@ -153,11 +153,11 @@ meeting.onaddstream = function(e) {
 };
 ```
 
-----
+=
 
 #### Browser Support
 
-This [WebRTC One-to-Many video-broadcasting](https://webrtc-experiment.appspot.com/one-to-many-video-broadcasting/) experiment works fine on following web-browsers:
+This [WebRTC One-to-Many video-broadcasting](https://www.webrtc-experiment.com/one-to-many-video-broadcasting/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -165,8 +165,8 @@ This [WebRTC One-to-Many video-broadcasting](https://webrtc-experiment.appspot.c
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[WebRTC One-to-Many video-broadcasting](https://webrtc-experiment.appspot.com/one-to-many-video-broadcasting/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC One-to-Many video-broadcasting](https://www.webrtc-experiment.com/one-to-many-video-broadcasting/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/one-to-many-video-broadcasting/index.html b/one-to-many-video-broadcasting/index.html
index 9ebab9df..eb1d06f2 100644
--- a/one-to-many-video-broadcasting/index.html
+++ b/one-to-many-video-broadcasting/index.html
@@ -108,8 +108,8 @@
                 padding-left: 1em;
             }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/one-to-many-video-broadcasting/meeting.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"></script>
+        <script src="https://www.webrtc-experiment.com/one-to-many-video-broadcasting/meeting.js"> </script>
     </head>
     <body>
         <h1>
@@ -120,7 +120,7 @@ <h1>
                                               Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <section>
             <h2>Setup a new meeting:</h2>
@@ -198,7 +198,7 @@ <h2>Difference between one-way broadcasting and one-to-many broadcasting</h2>
         <br />
         <br />
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/one-to-many-video-broadcasting/meeting.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/one-to-many-video-broadcasting/meeting.js"&gt;&lt;/script&gt;
 </pre>
         <pre>
 var meeting = new Meeting('meeting-unique-id');
@@ -245,8 +245,8 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
\ No newline at end of file
diff --git a/p2p-share/README.md b/p2p-share/README.md
index 8b1b7df9..719132bb 100644
--- a/p2p-share/README.md
+++ b/p2p-share/README.md
@@ -5,6 +5,8 @@ that allows you:
 1. Share files in a group (many-to-many)
 2. Unlimited data connections on Firefox
 
+=
+
 #### How WebRTC file sharing experiment works?
 
 1. It shares the file directly over all connected data ports
@@ -14,10 +16,14 @@ If 16 SCTP streams are used in one-to-one data connection. In 10 peers session;
 
 That's why it is strongly suggested to not open multiple peer connections on single page!
 
+=
+
 #### Is this a P2P file distributing system?
 
 Well, maybe!
 
+=
+
 #### Is this a torrent like file distributing/sharing system?
 
 This P2P file sharing experiment works like this:
@@ -27,6 +33,8 @@ This P2P file sharing experiment works like this:
 3. If `UserA` share file...file will be transferred asynchronously over all connected data ports.
 4. All other users are connected directly to each other; like a hexagon or other many directional shape.
 
+=
+
 #### Chrome and unreliable data connection...how it works?
 
 In a simple one-to-one data session; chrome opens two RTP ports:
@@ -36,9 +44,7 @@ In a simple one-to-one data session; chrome opens two RTP ports:
 
 So many limitations in the moment; to resolve all those limitations; data/files are splitted in small chunks; and those chunks are transferred **step-by-step** after predefined time interval to make distribution consistent and reliable.
 
-### How to use `One-to-Many File Sharing` in your own site?
-
-**Just copy HTML/JS code in your site and that's all you need to do. Nothing to install! No requirements!**
+=
 
 #### Browser Support
 
@@ -48,8 +54,9 @@ WebRTC **File Broadcasting** (file sharing) experiment works fine on following w
 | ------------- |-------------|
 | Firefox | [Stable](http://www.mozilla.org/en-US/firefox/new/) / [Aurora](http://www.mozilla.org/en-US/firefox/aurora/) / [Nightly](http://nightly.mozilla.org/) |
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
-| Internet Explorer / IE | [Chrome Frame](http://www.google.com/chromeframe) |
+
+=
 
 #### License
 
-WebRTC **File Broadcasting** experiment is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+WebRTC **File Broadcasting** experiment is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/part-of-screen-sharing/README.md b/part-of-screen-sharing/README.md
index 5af31bc2..94cf9b93 100644
--- a/part-of-screen-sharing/README.md
+++ b/part-of-screen-sharing/README.md
@@ -1,14 +1,16 @@
-#### WebRTC [Part of Screen Sharing Demos](https://webrtc-experiment.appspot.com/#part-of-screen-sharing)
+#### WebRTC [Part of Screen Sharing Demos](https://www.webrtc-experiment.com/#part-of-screen-sharing)
 
-1. [Using RTCDataChannel](https://webrtc-experiment.appspot.com/part-of-screen-sharing/webrtc-data-channel/)
-2. [Using Firebase](https://webrtc-experiment.appspot.com/part-of-screen-sharing/)
+1. [Using RTCDataChannel](https://www.webrtc-experiment.com/part-of-screen-sharing/webrtc-data-channel/)
+2. [Using Firebase](https://www.webrtc-experiment.com/part-of-screen-sharing/)
 3. [A realtime chat using RTCDataChannel](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/)
 4. [A realtime chat using Firebase](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/No-WebRTC-Chat.html)
 
+=
+
 #### How to use in your own site?
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/screenshot.js"></script>
+<script src="https://www.webrtc-experiment.com/screenshot.js"></script>
 ```
 
 ```javascript
@@ -37,6 +39,8 @@ html2canvas(divToShare, {
  */
 ```
 
+=
+
 #### How.....why.....what.....?
 
 1. Used `html2canvas` library to take screenshot of the entire webpage or part of webpage.
@@ -50,7 +54,7 @@ html2canvas(divToShare, {
 
 ....and that's all you need to do!
 
-**Just copy HTML/JS code in your site and that's all you need to do. Nothing to install! No requirements!**
+=
 
 #### Browser Support
 
@@ -63,6 +67,8 @@ These WebRTC **Part of Screen Sharing** experiments works fine on following web-
 | Firefox | [Nightly](http://nightly.mozilla.org/) |
 | Google Chrome | [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) |
 
+=
+
 #### License
 
-These WebRTC **Part of Screen Sharing** experiments are released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+These WebRTC **Part of Screen Sharing** experiments are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/part-of-screen-sharing/index.html b/part-of-screen-sharing/index.html
index 1f78712e..be5b796b 100644
--- a/part-of-screen-sharing/index.html
+++ b/part-of-screen-sharing/index.html
@@ -4,7 +4,7 @@
 <head>
     <title>Part of screen sharing ® Muaz Khan</title>
     <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
-    <meta charset=utf-8>
+    <meta charset="utf-8">
     <meta name="viewport" content="width=device-width, initial-scale=1.0, user-scalable=no">
     <link rel="author" type="text/html" href="https://plus.google.com/100325991024054712503">
     <meta name="author" content="Muaz Khan">
@@ -55,17 +55,17 @@
             text-align: center;
         }
 
-        footer a {
-            color: #666;
-            text-shadow: 0 1px 1px #fff;
-            text-decoration: none;
-            font-size: inherit;
-            padding: 1em;
-        }
+            footer a {
+                color: #666;
+                text-shadow: 0 1px 1px #fff;
+                text-decoration: none;
+                font-size: inherit;
+                padding: 1em;
+            }
 
-        footer a:hover, footer a:focus {
-            color: #111;
-        }
+                footer a:hover, footer a:focus {
+                    color: #111;
+                }
 
         h1, h2 {
             font-weight: normal;
@@ -80,13 +80,13 @@
             text-decoration: none;
         }
 
-        a:hover, a:focus {
-            color: #1B29A4;
-        }
+            a:hover, a:focus {
+                color: #1B29A4;
+            }
 
-        a:active {
-            color: #000;
-        }
+            a:active {
+                color: #000;
+            }
 
         :-moz-any-link:focus {
             color: #000;
@@ -120,20 +120,20 @@
             font-family: inherit;
         }
 
-        button:hover {
-            background: rgb(9, 147, 240);
-        }
+            button:hover {
+                background: rgb(9, 147, 240);
+            }
 
-        button:active {
-            background: rgb(10, 118, 190);
-        }
+            button:active {
+                background: rgb(10, 118, 190);
+            }
 
-        button[disabled] {
-            background: none;
-            border: 1px solid rgb(187, 181, 181);
-            color: gray;
-            text-shadow: none;
-        }
+            button[disabled] {
+                background: none;
+                border: 1px solid rgb(187, 181, 181);
+                color: gray;
+                text-shadow: none;
+            }
 
         strong {
             color: rgb(204, 14, 14);
@@ -149,9 +149,9 @@
             vertical-align: top;
         }
 
-        .share-me:hover {
-            background: rgb(211, 205, 205);
-        }
+            .share-me:hover {
+                background: rgb(211, 205, 205);
+            }
 
         #second-div {
             border: 2px solid red;
@@ -170,207 +170,205 @@
         document.createElement('article');
         document.createElement('footer');
     </script>
-    <script src="https://webrtc-experiment.appspot.com/screenshot.js"></script>
-    <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
+    <script src="https://www.webrtc-experiment.com/screenshot.js"></script>
+    <script src="https://www.webrtc-experiment.com/firebase.js"></script>
 </head>
 
 <body>
-<article>
-<a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+    <article>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none; right: 0; top: 0;">↑ WEBRTC EXPERIMENTS</a>
 
-<h1>
-    Part of screen sharing
+        <h1>Part of screen sharing
 
     <div title="Stop sharing your DIVs"
-         style="color:red;border:2px solid red;border-bottom:0;display:inline-block;padding:.1em .4em;">
+        style="color: red; border: 2px solid red; border-bottom: 0; display: inline-block; padding: .1em .4em;">
         <input type="checkbox" id="pause" checked>
         <label for="pause">Pause Sharing</label>
     </div>
 
-    <script>
-        var pause = document.getElementById('pause');
-        pause.onchange = function () {
-            isShareDIV = !this.checked;
-        };
-    </script>
-</h1>
-
-<p>
-    Copyright © 2013 <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a>&lt;<a
-        href="http://twitter.com/muazkh" target="_blank">@muazkh</a>&gt;.
-</p>
-<section class="plusone-gplus">
-    <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
-</section>
-<div id="first-div" class="share-me" contenteditable title="Tick checkbox to share this DIV">
-    <h2>First DIV</h2>
-
-    <br/>
-    <input type="checkbox" id="share-div-1">
-    <label for="share-div-1">Share DIV Number 1</label>
-</div>
-<div id="second-div" class="share-me" contenteditable title="Tick checkbox to share this DIV">
-    <h2>Second DIV</h2>
-
-    <br/>
-    <input type="checkbox" id="share-div-2">
-    <label for="share-div-2">Share DIV Number 2</label>
-</div>
-<div id="third-div" class="share-me" contenteditable title="Tick checkbox to share this DIV">
-    <h2>Third DIV</h2>
-
-    <br/>
-    <input type="checkbox" id="share-div-3">
-    <label for="share-div-3">Share DIV Number 3</label>
-</div>
-<script>
-    function find(id) {
-        return document.getElementById(id);
-    }
-    var shareFirstDIV = find('share-div-1'),
-            shareSecondDIV = find('share-div-2'),
-            shareThirdDIV = find('share-div-3');
-    shareFirstDIV.onchange = shareSecondDIV.onchange = shareThirdDIV.onchange = onchange;
-
-    function onchange(input) {
-        input = input || this;
-        var parent = input.parentNode;
-
-        if (input.checked) {
-            hide(true);
-            parent.style.display = 'inline-block';
-        } else hide(false);
-
-        if (input.checked) {
-            uncheckAll();
-            input.checked = true;
-        }
-
-        window.elementToShare = parent;
-        if (!isShareDIV) {
-            isShareDIV = true;
-            pause.checked = false;
-            shareDIV();
-        }
-
-        if (!input.checked) {
-            isShareDIV = false;
-            pause.checked = true;
-        }
-    }
+            <script>
+                var pause = document.getElementById('pause');
+                pause.onchange = function () {
+                    isShareDIV = !this.checked;
+                };
+            </script>
+        </h1>
+
+        <p>
+            Copyright © 2013 <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a>&lt;<a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>&gt;.
+            <a href="https://twitter.com/WebRTCWeb" target="_blank" title="WebRTC Experiments on Twitter">@WebRTCWeb</a>
+        </p>
+        <section class="plusone-gplus">
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
+        </section>
+        <div id="first-div" class="share-me" contenteditable title="Tick checkbox to share this DIV">
+            <h2>First DIV</h2>
+
+            <br />
+            <input type="checkbox" id="share-div-1">
+            <label for="share-div-1">Share DIV Number 1</label>
+        </div>
+        <div id="second-div" class="share-me" contenteditable title="Tick checkbox to share this DIV">
+            <h2>Second DIV</h2>
+
+            <br />
+            <input type="checkbox" id="share-div-2">
+            <label for="share-div-2">Share DIV Number 2</label>
+        </div>
+        <div id="third-div" class="share-me" contenteditable title="Tick checkbox to share this DIV">
+            <h2>Third DIV</h2>
+
+            <br />
+            <input type="checkbox" id="share-div-3">
+            <label for="share-div-3">Share DIV Number 3</label>
+        </div>
+        <script>
+            function find(id) {
+                return document.getElementById(id);
+            }
+            var shareFirstDIV = find('share-div-1'),
+                    shareSecondDIV = find('share-div-2'),
+                    shareThirdDIV = find('share-div-3');
+            shareFirstDIV.onchange = shareSecondDIV.onchange = shareThirdDIV.onchange = onchange;
+
+            function onchange(input) {
+                input = input || this;
+                var parent = input.parentNode;
+
+                if (input.checked) {
+                    hide(true);
+                    parent.style.display = 'inline-block';
+                } else hide(false);
+
+                if (input.checked) {
+                    uncheckAll();
+                    input.checked = true;
+                }
+
+                window.elementToShare = parent;
+                if (!isShareDIV) {
+                    isShareDIV = true;
+                    pause.checked = false;
+                    shareDIV();
+                }
+
+                if (!input.checked) {
+                    isShareDIV = false;
+                    pause.checked = true;
+                }
+            }
 
-    function uncheckAll() {
-        shareFirstDIV.checked =
-                shareSecondDIV.checked =
-                        shareThirdDIV.checked = false;
-    }
+            function uncheckAll() {
+                shareFirstDIV.checked =
+                        shareSecondDIV.checked =
+                                shareThirdDIV.checked = false;
+            }
 
-    function hide(isHide) {
-        shareFirstDIV.parentNode.style.display =
-                shareSecondDIV.parentNode.style.display =
-                        shareThirdDIV.parentNode.style.display = isHide ? 'none' : 'inline-block';
-    }
+            function hide(isHide) {
+                shareFirstDIV.parentNode.style.display =
+                        shareSecondDIV.parentNode.style.display =
+                                shareThirdDIV.parentNode.style.display = isHide ? 'none' : 'inline-block';
+            }
 
-    var shareMeDIVs = document.querySelectorAll('.share-me');
-    for (var i = 0; i < shareMeDIVs.length; i++) {
-        shareMeDIVs[i].onclick = function () {
-            var input = this.querySelector('input');
-            input.checked = true;
-            onchange(input);
-        };
-    }
-</script>
-<!-- this script is taking screenshot and transmitting them over firebase -->
-<script>
-    var isShareDIV = false;
-
-    function shareDIV() {
-        if (!isShareDIV) {
-            setTimeout(shareDIV, 1000);
-            return;
-        }
-        html2canvas(window.elementToShare, {
-            onrendered: function (canvas) {
-                var screenshot = canvas.toDataURL('image/webp', 1);
-                firebase && firebase.push({
-                    userID: userID,
-                    screenshot: screenshot
+            var shareMeDIVs = document.querySelectorAll('.share-me');
+            for (var i = 0; i < shareMeDIVs.length; i++) {
+                shareMeDIVs[i].onclick = function () {
+                    var input = this.querySelector('input');
+                    input.checked = true;
+                    onchange(input);
+                };
+            }
+        </script>
+        <!-- this script is taking screenshot and transmitting them over firebase -->
+        <script>
+            var isShareDIV = false;
+
+            function shareDIV() {
+                if (!isShareDIV) {
+                    setTimeout(shareDIV, 1000);
+                    return;
+                }
+                html2canvas(window.elementToShare, {
+                    onrendered: function (canvas) {
+                        var screenshot = canvas.toDataURL('image/webp', 1);
+                        firebase && firebase.push({
+                            userID: userID,
+                            screenshot: screenshot
+                        });
+                        //window.requestAnimationFrame(shareDIV);
+                        setTimeout(shareDIV, 500);
+                    }
                 });
-                //window.requestAnimationFrame(shareDIV);
-                setTimeout(shareDIV, 500);
             }
-        });
-    }
-</script>
-<br/>
-<br/>
+        </script>
+        <br />
+        <br />
 
-<div id="shared-parts-of-screen-preview" title="Your or your friend's shared DIVs are visible here.">
-    <br/>
-    <br/>
+        <div id="shared-parts-of-screen-preview" title="Your or your friend's shared DIVs are visible here.">
+            <br />
+            <br />
 
-    <h2>Shared DIVs will be visible here ↓</h2>
+            <h2>Shared DIVs will be visible here ↓</h2>
 
-</div>
-<!-- this script is getting screenshot shared with you -->
-<script>
-    var sharedPartsOfScreenPreview = document.getElementById('shared-parts-of-screen-preview');
+        </div>
+        <!-- this script is getting screenshot shared with you -->
+        <script>
+            var sharedPartsOfScreenPreview = document.getElementById('shared-parts-of-screen-preview');
 
-    function getNewImage(id) {
-        var image = new Image();
-        image.id = id;
+            function getNewImage(id) {
+                var image = new Image();
+                image.id = id;
 
-        sharedPartsOfScreenPreview.insertBefore(image, sharedPartsOfScreenPreview.firstChild);
-        return image;
-    }
+                sharedPartsOfScreenPreview.insertBefore(image, sharedPartsOfScreenPreview.firstChild);
+                return image;
+            }
 
-    var channel = location.hash.replace('#', '') || 'Part-of-Screen-Sharing';
-    var firebase = new Firebase('https://muazkh.firebaseIO.com/' + channel);
-    firebase.on("child_added", function (data) {
-        onMessage(data.val());
-    });
-    setTimeout(onFirebaseConnectionOpend, 1);
-    firebase.onDisconnect().remove();
-
-    function onMessage(data) {
-        if (!data.screenshot) return;
-        var image = document.getElementById(data.userID);
-        if (!image) image = getNewImage(data.userID);
-        image.src = data.screenshot;
-    }
+            var channel = location.hash.replace('#', '') || 'Part-of-Screen-Sharing';
+            var firebase = new Firebase('https://muazkh.firebaseIO.com/' + channel);
+            firebase.on("child_added", function (data) {
+                onMessage(data.val());
+            });
+            setTimeout(onFirebaseConnectionOpend, 1);
+            firebase.onDisconnect().remove();
+
+            function onMessage(data) {
+                if (!data.screenshot) return;
+                var image = document.getElementById(data.userID);
+                if (!image) image = getNewImage(data.userID);
+                image.src = data.screenshot;
+            }
 
-    window.userID = Math.random() * 1000 << 10;
+            window.userID = Math.random() * 1000 << 10;
 
-    function onFirebaseConnectionOpend() {
-        console.log('firebase connection is opened');
-    }
-</script>
-<br/>
-<br/>
-
-<h2>intro:</h2>
-
-<ol>
-    <li>Sharing <strong>part of the screen</strong> or <strong>region of screen</strong> (i.e. a DIV, SECTION, ARTICLE or ASIDE)... not the entire screen!</li>
-    <li>Everything inside that DIV is <strong>synchronized</strong> in realtime.</li>
-    <li>Works fine on all modern web browsers supporting WebRTC Data Channels.</li>
-</ol>
-<h2>how?</h2>
-
-<ol>
-    <li>Share screenshots of the entire DIV, ASIDE or BODY element in realtime!</li>
-    <li>Using Firebase to transmit screenshots in realtime! (sharing screenshots after each 500 milliseconds)</li>
-    <li>It is better to "<strong>pause sharing</strong>" to scroll down and see others' shared regions.</li>
-    <li>It supports multi-user connectivity too! Hmm! i.e. you can share region of screen with many friends!</li>
-</ol>
-<h2 style="text-align: center;font-weight: normal;display: block;"
-    title="Using RTCDataChannel APIs to share your DIVs' screenshots. It is super-fast and 100% realtime on Firefox!">
-    Try <a href="https://webrtc-experiment.appspot.com/part-of-screen-sharing/webrtc-data-channel/" target="_blank">Part
+            function onFirebaseConnectionOpend() {
+                console.log('firebase connection is opened');
+            }
+        </script>
+        <br />
+        <br />
+
+        <h2>intro:</h2>
+
+        <ol>
+            <li>Sharing <strong>part of the screen</strong> or <strong>region of screen</strong> (i.e. a DIV, SECTION, ARTICLE or ASIDE)... not the entire screen!</li>
+            <li>Everything inside that DIV is <strong>synchronized</strong> in realtime.</li>
+            <li>Works fine on all modern web browsers supporting WebRTC Data Channels.</li>
+        </ol>
+        <h2>how?</h2>
+
+        <ol>
+            <li>Share screenshots of the entire DIV, ASIDE or BODY element in realtime!</li>
+            <li>Using Firebase to transmit screenshots in realtime! (sharing screenshots after each 500 milliseconds)</li>
+            <li>It is better to "<strong>pause sharing</strong>" to scroll down and see others' shared regions.</li>
+            <li>It supports multi-user connectivity too! Hmm! i.e. you can share region of screen with many friends!</li>
+        </ol>
+        <h2 style="text-align: center; font-weight: normal; display: block;"
+            title="Using RTCDataChannel APIs to share your DIVs' screenshots. It is super-fast and 100% realtime on Firefox!">Try <a href="https://www.webrtc-experiment.com/part-of-screen-sharing/webrtc-data-channel/" target="_blank">Part
     of Screen Sharing using WebRTC Data Channel</a> / 100% realtime!
-</h2>
+        </h2>
 
-            <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/screenshot.js"&gt;&lt;/script&gt;
+        <pre>
+&lt;script src="https://www.webrtc-experiment.com/screenshot.js"&gt;&lt;/script&gt;
 &lt;script&gt;
 var divToShare = document.querySelector('div');
 html2canvas(divToShare, {
@@ -398,38 +396,34 @@ <h2>What above code will do?</h2>
 &lt;/script&gt;
             </pre>
 
-<br/>
-<br/>
+        <br />
+        <br />
 
-<h2 style="display: block;text-align: center;"><a
-        href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing" target="_blank">Part of
+        <h2 style="display: block; text-align: center;"><a
+            href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing" target="_blank">Part of
     screen sharing is open-sourced on Github!</a></h2>
 
-<br/>
-<br/>
-<table style="width: 100%;">
-    <tr>
-        <td>
+        <br />
+        <br />
+        <section style="border: 1px solid rgb(189, 189, 189); margin: 1em 3em; border-radius: .2em;">
+            <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 189, 189);">Feedback</h2>
             <div>
-                <textarea id="message" style="height: 8em; width: 100%;"
-                          placeholder="Have any message? Suggestions or something went wrong?"></textarea>
+                <textarea id="message" style="height: 8em; margin: .2em; width: 98%; border: 1px solid rgb(189, 189, 189); outline: none; resize: vertical;" placeholder="Have any message? Suggestions or something went wrong?"></textarea>
             </div>
-            <button id="send-message" class="send-message" style="font-size: 1em;">Send Message</button>
-        </td>
-    </tr>
-</table>
-</article>
-<footer>
-    <p>
-        <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
-        <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
-        <span> 2013 </span> »
-        <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
-        <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
-        <a href="https://github.com/muaz-khan" target="_blank">Github</a>
-    </p>
-</footer>
-<script src="https://webrtc-experiment.appspot.com/common.js"></script>
+            <button id="send-message" style="font-size: 1em;">Send Message</button>
+        </section>
+    </article>
+    <footer>
+        <p>
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
+                <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
+                <span>2013 </span>»
+                <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+                <a href="https://twitter.com/muazkh" target="_blank">@muazkh</a>»
+                <a href="https://twitter.com/WebRTCWeb" target="_blank">@WebRTCWeb</a>»
+                <a href="https://github.com/muaz-khan" target="_blank">Github</a>
+        </p>
+    </footer>
+    <script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
-
-</html>
\ No newline at end of file
+</html>
diff --git a/part-of-screen-sharing/realtime-chat/README.md b/part-of-screen-sharing/realtime-chat/README.md
index 1d10d13d..5776b14d 100644
--- a/part-of-screen-sharing/realtime-chat/README.md
+++ b/part-of-screen-sharing/realtime-chat/README.md
@@ -1,13 +1,13 @@
-====
-## WebRTC Part of Screen Sharing Demos
+#### WebRTC Part of Screen Sharing Demos
 
 1. [Realtime Chat](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/)
 2. [No-WebRTC Realtime Chat](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/No-WebRTC-Chat.html)
 3. [Part of Screen Sharing](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/part-of-screen-sharing/)
 4. [Part of Screen Sharing using RTCDataChannel](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/part-of-screen-sharing/RTCDataChannel/)
 
-====
-## Browser Support
+=
+
+#### Browser Support
 
 WebRTC [Part of Screen Sharing using RTCDataChannel](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/part-of-screen-sharing/RTCDataChannel/) experiment works fine on following web-browsers:
 
@@ -18,7 +18,8 @@ WebRTC [Part of Screen Sharing using RTCDataChannel](https://googledrive.com/hos
 | Firefox | [Nightly](http://nightly.mozilla.org/) |
 | Google Chrome | [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) |
 
-====
-## License
+=
+
+#### License
 
-These WebRTC **Part of Screen Sharing** experiments are released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+These WebRTC **Part of Screen Sharing** experiments are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/part-of-screen-sharing/realtime-chat/how-this-work.html b/part-of-screen-sharing/realtime-chat/how-this-work.html
index b68cd346..5ad17721 100644
--- a/part-of-screen-sharing/realtime-chat/how-this-work.html
+++ b/part-of-screen-sharing/realtime-chat/how-this-work.html
@@ -27,7 +27,7 @@ <h1>
 		</h1>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <ol>
             <li>Tick "<code>Is Sync in Realtime</code>" checkbox if you want to share text in realtime.</li>
diff --git a/part-of-screen-sharing/realtime-chat/index.html b/part-of-screen-sharing/realtime-chat/index.html
index 9420ffc3..34e3113b 100644
--- a/part-of-screen-sharing/realtime-chat/index.html
+++ b/part-of-screen-sharing/realtime-chat/index.html
@@ -64,7 +64,7 @@ <h1>
 		</h1>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <input type="checkbox" id="realtime" checked>
         <label for="realtime">Is Sync in Realtime?</label>/
diff --git a/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/README.md b/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/README.md
index 5af31bc2..94cf9b93 100644
--- a/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/README.md
+++ b/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/README.md
@@ -1,14 +1,16 @@
-#### WebRTC [Part of Screen Sharing Demos](https://webrtc-experiment.appspot.com/#part-of-screen-sharing)
+#### WebRTC [Part of Screen Sharing Demos](https://www.webrtc-experiment.com/#part-of-screen-sharing)
 
-1. [Using RTCDataChannel](https://webrtc-experiment.appspot.com/part-of-screen-sharing/webrtc-data-channel/)
-2. [Using Firebase](https://webrtc-experiment.appspot.com/part-of-screen-sharing/)
+1. [Using RTCDataChannel](https://www.webrtc-experiment.com/part-of-screen-sharing/webrtc-data-channel/)
+2. [Using Firebase](https://www.webrtc-experiment.com/part-of-screen-sharing/)
 3. [A realtime chat using RTCDataChannel](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/)
 4. [A realtime chat using Firebase](https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/realtime-chat/No-WebRTC-Chat.html)
 
+=
+
 #### How to use in your own site?
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/screenshot.js"></script>
+<script src="https://www.webrtc-experiment.com/screenshot.js"></script>
 ```
 
 ```javascript
@@ -37,6 +39,8 @@ html2canvas(divToShare, {
  */
 ```
 
+=
+
 #### How.....why.....what.....?
 
 1. Used `html2canvas` library to take screenshot of the entire webpage or part of webpage.
@@ -50,7 +54,7 @@ html2canvas(divToShare, {
 
 ....and that's all you need to do!
 
-**Just copy HTML/JS code in your site and that's all you need to do. Nothing to install! No requirements!**
+=
 
 #### Browser Support
 
@@ -63,6 +67,8 @@ These WebRTC **Part of Screen Sharing** experiments works fine on following web-
 | Firefox | [Nightly](http://nightly.mozilla.org/) |
 | Google Chrome | [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) |
 
+=
+
 #### License
 
-These WebRTC **Part of Screen Sharing** experiments are released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+These WebRTC **Part of Screen Sharing** experiments are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/index.html b/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/index.html
index 11fb4869..41ece8fc 100644
--- a/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/index.html
+++ b/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/index.html
@@ -9,6 +9,13 @@
     <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>
+            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>
     <style>
         @import url(https://fonts.googleapis.com/css?family=Inconsolata);
 
@@ -170,14 +177,14 @@
         document.createElement('article');
         document.createElement('footer');
     </script>
-    <script src="https://webrtc-experiment.appspot.com/DataChannel.js"> </script>
-    <script src="https://webrtc-experiment.appspot.com/screenshot.js"> </script>
-    <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
+    <script src="https://www.webrtc-experiment.com/DataChannel.js"> </script>
+    <script src="https://www.webrtc-experiment.com/screenshot.js"> </script>
+    <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
 </head>
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; right: 0; text-decoration: none; top: 0;">↑ WEBRTC EXPERIMENTS</a>
 
         <h1>
             <div title="Stop sharing your DIVs"
@@ -201,7 +208,7 @@ <h1>
                 <span style="color: rgb(153, 34, 34); font-weight: bold; background: rgb(247, 247, 247); padding: .2em .4em; border: 2px solid rgb(136, 67, 67);">Firefox Nightly is recommended for this experiment.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <div id="first-div" class="share-me" contenteditable title="Tick checkbox to share this DIV">
@@ -227,8 +234,10 @@ <h2>Third DIV</h2>
         </div>
 
         <script>
-            // using DataChannel.js library for WebRTC Data Channels
-            var channel = new DataChannel(location.hash.replace('#', '') || 'part-of-screen-sharing');
+			// using DataChannel.js library for WebRTC Data Channels
+			var channel = new DataChannel(location.hash.substr(1) || 'part-of-screen-sharing', {
+				firebase: 'webrtc'
+			});
 
             channel.onopen = function () {
                 window.IsDataChannelOpened = true;
@@ -379,12 +388,12 @@ <h2>how?</h2>
             <li>It supports multi-user connectivity too! Hmm! i.e. you can share region of screen with many friends!</li>
         </ol>
         <h2 style="display: block; font-weight: normal; text-align: center;"
-            title="Using RTCDataChannel APIs to share your DIVs' screenshots. It is super-fast and 100% realtime on Firefox!">Try <a href="https://webrtc-experiment.appspot.com/part-of-screen-sharing/" target="_blank">Part
+            title="Using RTCDataChannel APIs to share your DIVs' screenshots. It is super-fast and 100% realtime on Firefox!">Try <a href="https://www.webrtc-experiment.com/part-of-screen-sharing/" target="_blank">Part
                         of Screen Sharing using Firebase</a>
         </h2>
 
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/screenshot.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/screenshot.js"&gt;&lt;/script&gt;
 &lt;script&gt;
 var divToShare = document.querySelector('div');
 html2canvas(divToShare, {
@@ -431,7 +440,7 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>,
                 <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
@@ -440,6 +449,6 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/realtime-pluginfree-calls/README.md b/realtime-pluginfree-calls/README.md
index 6db58dfc..08d6e876 100644
--- a/realtime-pluginfree-calls/README.md
+++ b/realtime-pluginfree-calls/README.md
@@ -1,4 +1,4 @@
-##### Browser-to-Browser audio-only calls / [Demo](https://webrtc-experiment.appspot.com/calls/)
+##### Browser-to-Browser audio-only calls / [Demo](https://www.webrtc-experiment.com/calls/)
 
 It is one-to-one audio-sharing experiment. You'll be asked to allow access to microphone or audio input device to establish voice-only call between you and your friend.
 
@@ -12,7 +12,7 @@ Features:
 
 ##### Browser Support
 
-[Audio-Only calling experiment](https://webrtc-experiment.appspot.com/calls/) works fine on following web-browsers:
+[Audio-Only calling experiment](https://www.webrtc-experiment.com/calls/) works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -24,4 +24,4 @@ Features:
 
 #### License
 
-[Audio-Only calling experiment](https://webrtc-experiment.appspot.com/calls/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[Audio-Only calling experiment](https://www.webrtc-experiment.com/calls/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/realtime-pluginfree-calls/index.html b/realtime-pluginfree-calls/index.html
index 800208d7..f47be746 100644
--- a/realtime-pluginfree-calls/index.html
+++ b/realtime-pluginfree-calls/index.html
@@ -142,13 +142,13 @@
     </head>
     
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
-             <h1>WebRTC Calls: Realtime & Pluginfree!</h1>
+             <h1>WebRTC Calls: Realtime & Pluginfree! / <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/realtime-pluginfree-calls" target="_blank">Source Code on Github</a></h1>
 
             <p> <span>Copyright © 2013</span> <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span></p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <table class="visible">
                 <tr>
@@ -178,17 +178,13 @@ <h2>
                 </tr>
             </table>
             <div id="participants"></div>
-            <script src='https://webrtc-experiment.appspot.com/firebase.js'></script>
-            <script src="https://webrtc-experiment.appspot.com/lib/RTCPeerConnection-v1.4.js"></script>
-            <script src="https://webrtc-experiment.appspot.com/RecordRTC.js"></script>
-            <script src="https://webrtc-experiment.appspot.com/calls/call-initiator.js"></script>
-            <script src="https://webrtc-experiment.appspot.com/calls/ui.js"></script>
-            <br />
-            <br />
-             <h2 style="float:right;"><a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/realtime-pluginfree-calls" target="_blank">Source Code on Github</a></h2>
-
-            <br />
-            <br />
+            <script src='https://www.webrtc-experiment.com/firebase.js'></script>
+            <script src="https://www.webrtc-experiment.com/lib/RTCPeerConnection-v1.4.js"></script>
+            <script src="https://www.webrtc-experiment.com/RecordRTC.js"></script>
+            <script src="https://www.webrtc-experiment.com/calls/call-initiator.js"></script>
+            <script src="https://www.webrtc-experiment.com/calls/ui.js"></script>
+        </article>
+		<article>
              <h2>Features</h2>
 
             <ol>
@@ -196,21 +192,20 @@ <h2>Features</h2>
                 <li>Cross-Browser calling</li>
                 <li>Voice-Recording</li>
             </ol>
-            <br />
-            <br />
-            	<h2>Try <a href="https://webrtc-experiment.appspot.com/RTCall/" target="_blank">RTCall.js</a> for Browser-to-Browser audio-only calling</h2>
+        </article>
+		<article>
+            <h2>Try <a href="https://www.webrtc-experiment.com/RTCall/" target="_blank">RTCall.js</a> for Browser-to-Browser audio-only calling</h2>
 
             <ol>
                 <li>In your own site; as an admin; you may want to be auto-alerted for each new visitor i.e. customer.</li>
                 <li>You may want to make voice-only calls between admin and customers</li>
             </ol>	<a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCall" target="_blank">RTCall.js Documentation & Source code</a>
 
-            <br />
-            <br />
+        </article>
+		<article>
             <p>You can use <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection" target="_blank">RTCMultiConnection.js</a> for audio/video/screen sharing (i.e. conferencing/broadcasting); it supports multi-sessions establishment on the
                 same page, group text chat and file sharing, auto user presence detection and many other features.</p>
-            <br />
-            <br />
+            
             <section style="border: 1px solid rgb(189, 189, 189); margin: 1em 3em; border-radius: .2em;">
                  <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 189, 189);">Feedback</h2>
 
@@ -221,13 +216,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                 href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
 
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+        <script src="https://www.webrtc-experiment.com/common.js"></script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/screen-broadcast/README.md b/screen-broadcast/README.md
index 85b2969d..78a5b364 100644
--- a/screen-broadcast/README.md
+++ b/screen-broadcast/README.md
@@ -2,10 +2,14 @@
 
 Sharing tab using chrome **experimental tabCapture APIs**; broadcasting over many peers.
 
-#### [You can view broadcasted tabs here](https://webrtc-experiment.appspot.com/screen-broadcast/)
+=
+
+#### [You can view broadcasted tabs here](https://www.webrtc-experiment.com/screen-broadcast/)
 
 You can also view broadcasted tab using Firefox nightly, aurora, and 18+stable! It is cross-browser!
 
+=
+
 #### How to capture stream using tabCapture APIs?
 
 ```javascript
@@ -26,12 +30,16 @@ chrome.tabs.getSelected(null, function (tab) {
 });
 ```
 
+=
+
 #### How to use your own socket.io implementation?
 
 This tab sharing experiment is using socket.io implementation of pubnub.
 
 1. broadcast.js � see in [tabCapture extension](http://code.google.com/p/muazkh/downloads/list)
-2. screen-viewer.js � see in [screen-broadcast](https://webrtc-experiment.appspot.com/screen-broadcast/)
+2. screen-viewer.js � see in [screen-broadcast](https://www.webrtc-experiment.com/screen-broadcast/)
+
+=
 
 ##### broadcast.js � Line 181
 
@@ -51,6 +59,8 @@ openSocket: function(config) {
 }
 ```
 
+=
+
 ##### screen-viewer.js � Line 185
 
 At line `185`, you can see **same** `openSocket` method:
@@ -69,6 +79,8 @@ openSocket: function(config) {
 }
 ```
 
+=
+
 ##### To use your own socket.io implementation...
 
 Replace `openSocket` method's code in **both files** with:
@@ -90,10 +102,14 @@ openSocket: function (config) {
 
 Now, **your own socket.io will be used for signaling!**
 
+=
+
 ##### Keep in mind....
 
 Keep in mind that you **must** link appropriate `socket.io.js` file.
 
+=
+
 #### Is it possible to share full screen using tabCapture APIs?
 
 Maybe. It seems that [CaptureInfo](http://developer.chrome.com/dev/extensions/tabCapture.html#type-CaptureInfo) object has `fullscreen` boolean parameter which let us know whether **an element in the tab being captured is in fullscreen mode**.
@@ -127,6 +143,8 @@ chrome.tabCapture.capture(constraints, function (stream) {
 
 Well, it is suggested to try **fullscreen mode** yourself. Maybe it work!
 
+=
+
 #### Browser support of tabCapture APIs
 
 From April, 2013:
@@ -136,6 +154,8 @@ From April, 2013:
 | Google Chrome | [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) |
 | Google Chrome | [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 
+=
+
 #### List of browsers that can view broadcasted tab
 
 | Browser        | Support           |
@@ -144,10 +164,14 @@ From April, 2013:
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Internet Explorer / IE | [Chrome Frame](http://www.google.com/chromeframe) |
 
+=
+
 #### Plugin-free screen sharing
 
 There is a plugin-free screen sharing experiment too! [Try it Now!](https://googledrive.com/host/0B6GWd_dUUTT8WHpWSzZ5S0RqeUk/Pluginfree-Screen-Sharing.html)
 
+=
+
 #### License
 
-[TabCapture Extension](http://code.google.com/p/muazkh/downloads/list) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[TabCapture Extension](http://code.google.com/p/muazkh/downloads/list) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/screen-broadcast/index.html b/screen-broadcast/index.html
index b4e02b84..49b870af 100644
--- a/screen-broadcast/index.html
+++ b/screen-broadcast/index.html
@@ -157,7 +157,7 @@
 
     <body>
         <article>
-            <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+            <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
             <h1>WebRTC Tab Sharing / Broadcasting</h1>
             <p>
@@ -166,7 +166,7 @@ <h1>WebRTC Tab Sharing / Broadcasting</h1>
             </p>
 
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
 		
             <table id="rooms-list" class="visible">
@@ -182,9 +182,9 @@ <h1>WebRTC Tab Sharing / Broadcasting</h1>
 
             <div id="participants"></div>
 
-            <script src="https://webrtc-experiment.appspot.com/dependencies/socket.io.js"></script>
-            <script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"> </script>
-            <script src="https://webrtc-experiment.appspot.com/screen-broadcast/screen-viewer.js"> </script>
+            <script src="https://www.webrtc-experiment.com/dependencies/socket.io.js"></script>
+            <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"> </script>
+            <script src="https://www.webrtc-experiment.com/screen-broadcast/screen-viewer.js"> </script>
 
             <br />
             <br />
@@ -200,10 +200,10 @@ <h1>WebRTC Tab Sharing / Broadcasting</h1>
             <blockquote>
                 <ol>
                     <li>
-                        <a href="https://webrtc-experiment.appspot.com/screen-broadcast/how-to-install/">How to install tabCapture extension?</a>
+                        <a href="https://www.webrtc-experiment.com/screen-broadcast/how-to-install/">How to install tabCapture extension?</a>
                     </li>
                     <li>
-                        <a href="https://webrtc-experiment.appspot.com/howto/broadcast-screen/">How to share tab using RTCPeerConnection APIs?</a>
+                        <a href="https://www.webrtc-experiment.com/howto/broadcast-screen/">How to share tab using RTCPeerConnection APIs?</a>
                     </li>
                 </ol>
             </blockquote>
@@ -246,13 +246,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         </article>
         <footer>
             <p>
-                <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+                <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
\ No newline at end of file
diff --git a/screen-sharing/README.md b/screen-sharing/README.md
index 0f265141..d28dcdb7 100644
--- a/screen-sharing/README.md
+++ b/screen-sharing/README.md
@@ -1,7 +1,9 @@
-#### WebRTC Screen-Sharing / [Demo](https://webrtc-experiment.appspot.com/screen-sharing/)
+#### WebRTC Screen-Sharing / [Demo](https://www.webrtc-experiment.com/screen-sharing/)
 
 Ad-ons free; plugin-free; extension free; direct browser-to-browser screen sharing.
 
+=
+
 #### Enable screen capture support in getUserMedia()
 
 You must enable this flag via `chrome://flags`.
@@ -19,7 +21,7 @@ navigator.webkitGetUserMedia({
 }, onsuccess, onfailure);
 ```
 
-----
+=
 
 #### Desktop Sharing?
 
@@ -27,7 +29,7 @@ Obviously, it is one of the most requested features; however not supported yet.
 
 These screen sharing APIs (i.e. `{ chromeMediaSource: 'screen' }`) allows only state-less (non-interactive) screen sharing.
 
-----
+=
 
 #### To use code in your own site, you must understand following limitations:
 
@@ -38,23 +40,23 @@ Chrome canary denies **screen capturing** request automatically if:
 3. You've not installed SSL certificate (i.e. testing on non-HTTPS domain)
 4. **screen capturing** is requested multiple times per tab. Maximum one request is permitted per page!
 
-----
+=
 
 #### Why recursive cascade images or blurred screen?
 
 Remember, recursive cascade images or blurred screen is chrome's implementation issue. It will be solved soon.
 
-`mandatory: {chromeMediaSource: 'tab'}` can only be useful in chrome extensions. See [Tab sharing using tabCapture APIs](https://webrtc-experiment.appspot.com/screen-broadcast/).
+`mandatory: {chromeMediaSource: 'tab'}` can only be useful in chrome extensions. See [Tab sharing using tabCapture APIs](https://www.webrtc-experiment.com/screen-broadcast/).
 
-----
+=
 
 #### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/screen-sharing/screen.js"></script>
+<script src="https://www.webrtc-experiment.com/screen-sharing/screen.js"></script>
 ```
 
-----
+=
 
 #### Last Step: Start using it!
 
@@ -76,7 +78,7 @@ document.getElementById('share-screen').onclick = function() {
 };
 ```
 
-----
+=
 
 #### Custom user-ids?
 
@@ -84,7 +86,7 @@ document.getElementById('share-screen').onclick = function() {
 screen.userid = 'username';
 ```
 
-----
+=
 
 #### Custom signaling channel?
 
@@ -121,7 +123,7 @@ Want to use `Firebase` for signaling?
 screen.firebase = 'chat';
 ```
 
-----
+=
 
 #### Want to manually join rooms?
 
@@ -142,7 +144,7 @@ screen.onscreen = function(_screen) {
 
 `onscreen` is called for each new screen; and `view` method allows you manually view shared screens.
 
-----
+=
 
 #### If someone leaves...
 
@@ -156,7 +158,7 @@ screen.onuserleft = function(userid) {
 };
 ```
 
-----
+=
 
 #### `onaddstream`
 
@@ -173,11 +175,11 @@ screen.onaddstream = function(e) {
 };
 ```
 
-----
+=
 
 #### Browser Support
 
-This [WebRTC Screen Sharing](https://webrtc-experiment.appspot.com/screen-sharing/) experiment works fine on following web-browsers:
+This [WebRTC Screen Sharing](https://www.webrtc-experiment.com/screen-sharing/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -185,8 +187,8 @@ This [WebRTC Screen Sharing](https://webrtc-experiment.appspot.com/screen-sharin
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[WebRTC Screen Sharing](https://webrtc-experiment.appspot.com/screen-sharing/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC Screen Sharing](https://www.webrtc-experiment.com/screen-sharing/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/screen-sharing/index.html b/screen-sharing/index.html
index 365b6d5c..07fe8e1c 100644
--- a/screen-sharing/index.html
+++ b/screen-sharing/index.html
@@ -111,8 +111,8 @@
                 padding-left: 1em;
             }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/screen-sharing/screen.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/screen-sharing/screen.js"> </script>
     </head>
     <body>
         <h1>
@@ -124,7 +124,7 @@ <h1>
                                               Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <section>
             <h2>Share Your Screen:</h2>
@@ -183,7 +183,7 @@ <h2>
         <br />
         <br />
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/screen-sharing/screen.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/screen-sharing/screen.js"&gt;&lt;/script&gt;
 </pre>
         <pre>
 var screen = new Screen('screen-unique-id');
@@ -240,7 +240,7 @@ <h2>
         </p>
         <p>
             mandatory: {chromeMediaSource: 'tab'} can only be useful in chrome extensions. See
-            <a href="https://webrtc-experiment.appspot.com/screen-broadcast/" target="_blank">Tab
+            <a href="https://www.webrtc-experiment.com/screen-broadcast/" target="_blank">Tab
                 sharing using tabCapture APIs.</a></p>
         <br />
         <br />
@@ -263,8 +263,8 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
\ No newline at end of file
diff --git a/socket.io/README.md b/socket.io/README.md
index d2873ce2..edcf933c 100644
--- a/socket.io/README.md
+++ b/socket.io/README.md
@@ -1,8 +1,8 @@
-#### WebRTC One-to-One video sharing using Socket.io / [Demo](https://webrtc-experiment.appspot.com/socket.io/)
+#### WebRTC One-to-One video sharing using Socket.io / [Demo](https://www.webrtc-experiment.com/socket.io/)
 
 This `WebRTC Experiment` is using [socket.io over node.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socketio-over-nodejs) for signalig.
 
-----
+=
 
 #### Use your own socket.io implementation!
 
@@ -38,11 +38,11 @@ openSignalingChannel: function(config) {
 }
 ```
 
-----
+=
 
 #### Browser Support
 
-This [One-to-one WebRTC video chat using socket.io](https://webrtc-experiment.appspot.com/socket.io/) experiment works fine on following web-browsers:
+This [One-to-one WebRTC video chat using socket.io](https://www.webrtc-experiment.com/socket.io/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -50,8 +50,8 @@ This [One-to-one WebRTC video chat using socket.io](https://webrtc-experiment.ap
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[WebRTC one-to-one video sharing using socket.io](https://webrtc-experiment.appspot.com/socket.io/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC one-to-one video sharing using socket.io](https://www.webrtc-experiment.com/socket.io/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/socket.io/index.html b/socket.io/index.html
index e4534264..a09f9ada 100644
--- a/socket.io/index.html
+++ b/socket.io/index.html
@@ -172,7 +172,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
         <h1>Using socket.io for signaling / <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socket.io" target="_blank">Source Code on Github</a></h1>
         <p>
@@ -181,7 +181,7 @@ <h1>Using socket.io for signaling / <a href="https://github.com/muaz-khan/WebRTC
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table class="visible">
@@ -208,10 +208,10 @@ <h2>
 
         <div id="participants"></div>
 
-        <script src="https://webrtc-experiment.appspot.com/dependencies/socket.io.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/socket.io/rtclib.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/socket.io/ui.js"></script>
+        <script src="https://www.webrtc-experiment.com/socket.io.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
+        <script src="https://www.webrtc-experiment.com/socket.io/rtclib.js"> </script>
+        <script src="https://www.webrtc-experiment.com/socket.io/ui.js"></script>
 		<p>WebRTC one-to-one video sharing experiment where socket.io is used for signaling.</p>
     </article>    
     <article>
@@ -236,13 +236,13 @@ <h2>Use <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/sock
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/socket.io/rtclib.js b/socket.io/rtclib.js
index 82eba9d7..4b00d3d4 100644
--- a/socket.io/rtclib.js
+++ b/socket.io/rtclib.js
@@ -1,21 +1,25 @@
-/* MIT License: https://webrtc-experiment.appspot.com/licence/ */
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing
 
-var rtclib = function (config) {
+var RTCLib = function(config) {
     var self = {
         userToken: uniqueToken()
-    },
-        channels = '--',
-        isbroadcaster,
+    };
+
+    var isbroadcaster,
         isGetNewRoom = true,
-        defaultSocket = {};
+        sockets = [],
+        isGotRemoteStream = false,
+        defaultSocket = { };
 
     function openDefaultSocket() {
         defaultSocket = config.openSocket({
-                onmessage: onDefaultSocketResponse,
-                callback: function (socket) {
-                    defaultSocket = socket;
-                }
-            });
+            onmessage: onDefaultSocketResponse,
+            callback: function(socket) {
+                defaultSocket = socket;
+            }
+        });
     }
 
     function onDefaultSocketResponse(response) {
@@ -25,13 +29,11 @@ var rtclib = function (config) {
 
         if (response.newParticipant) onNewParticipant(response.newParticipant);
 
-        if (response.userToken && response.joinUser == self.userToken && response.participant && channels.indexOf(response.userToken) == -1) {
-            channels += response.userToken + '--';
+        if (response.userToken && response.joinUser == self.userToken && response.participant) {
             openSubSocket({
-                    isofferer: true,
-                    channel: response.channel || response.userToken,
-                    closeSocket: true
-                });
+                isofferer: true,
+                channel: response.channel || response.userToken
+            });
         }
     }
 
@@ -40,35 +42,35 @@ var rtclib = function (config) {
         var socketConfig = {
             channel: _config.channel,
             onmessage: socketResponse,
-            onopen: function () {
+            onopen: function() {
                 if (isofferer && !peer) initPeer();
+                sockets[sockets.length] = socket;
             }
         };
 
-        socketConfig.callback = function (_socket) {
+        socketConfig.callback = function(_socket) {
             socket = _socket;
             this.onopen();
         };
 
         var socket = config.openSocket(socketConfig),
             isofferer = _config.isofferer,
-            gotstream,
             video = document.createElement('video'),
-            inner = {},
+            inner = { },
             peer;
 
         var peerConfig = {
             attachStream: config.attachStream,
-            onICE: function (candidate) {
+            onICE: function(candidate) {
                 socket.send({
-                        userToken: self.userToken,
-                        candidate: {
-                            sdpMLineIndex: candidate.sdpMLineIndex,
-                            candidate: JSON.stringify(candidate.candidate)
-                        }
-                    });
+                    userToken: self.userToken,
+                    candidate: {
+                        sdpMLineIndex: candidate.sdpMLineIndex,
+                        candidate: JSON.stringify(candidate.candidate)
+                    }
+                });
             },
-            onRemoteStream: function (stream) {
+            onRemoteStream: function(stream) {
                 if (!stream) return;
 
                 video[moz ? 'mozSrcObject' : 'src'] = moz ? stream : webkitURL.createObjectURL(stream);
@@ -76,6 +78,12 @@ var rtclib = function (config) {
 
                 _config.stream = stream;
                 onRemoteStreamStartsFlowing();
+            },
+            onRemoteStreamEnded: function(stream) {
+                if (config.onRemoteStreamEnded)
+                    config.onRemoteStreamEnded(stream);
+
+                isGotRemoteStream = false;
             }
         };
 
@@ -92,80 +100,45 @@ var rtclib = function (config) {
 
         function onRemoteStreamStartsFlowing() {
             if (!(video.readyState <= HTMLMediaElement.HAVE_CURRENT_DATA || video.paused || video.currentTime <= 0)) {
-                gotstream = true;
+                isGotRemoteStream = true;
 
-                config.onRemoteStream({
+                if (config.onRemoteStream)
+                    config.onRemoteStream({
                         video: video,
                         stream: _config.stream
                     });
-
-                if (isbroadcaster && channels.split('--').length > 3) {
-                    /* broadcasting newly connected participant for video-conferencing! */
-                    defaultSocket.send({
-                            newParticipant: socket.channel,
-                            userToken: self.userToken
-                        });
-                }
-
-                /* closing subsocket here on the offerer side */
-                if (_config.closeSocket) socket = null;
-
             } else setTimeout(onRemoteStreamStartsFlowing, 50);
         }
 
         function sendsdp(sdp) {
-            sdp = JSON.stringify(sdp);
-            var part = parseInt(sdp.length / 3);
-
-            var firstPart = sdp.slice(0, part),
-                secondPart = sdp.slice(part, sdp.length - 1),
-                thirdPart = '';
-
-            if (sdp.length > part + part) {
-                secondPart = sdp.slice(part, part + part);
-                thirdPart = sdp.slice(part + part, sdp.length);
-            }
-
-            socket.send({
-                    userToken: self.userToken,
-                    firstPart: firstPart
-                });
-
-            socket.send({
-                    userToken: self.userToken,
-                    secondPart: secondPart
-                });
-
             socket.send({
-                    userToken: self.userToken,
-                    thirdPart: thirdPart
-                });
+                userToken: self.userToken,
+                sdp: JSON.stringify(sdp)
+            });
         }
 
         function socketResponse(response) {
             if (response.userToken == self.userToken) return;
-            if (response.firstPart || response.secondPart || response.thirdPart) {
-                if (response.firstPart) {
-                    inner.firstPart = response.firstPart;
-                    if (inner.secondPart && inner.thirdPart) selfInvoker();
-                }
-                if (response.secondPart) {
-                    inner.secondPart = response.secondPart;
-                    if (inner.firstPart && inner.thirdPart) selfInvoker();
-                }
-
-                if (response.thirdPart) {
-                    inner.thirdPart = response.thirdPart;
-                    if (inner.firstPart && inner.secondPart) selfInvoker();
-                }
+            if (response.sdp) {
+                inner.sdp = JSON.parse(response.sdp);
+                selfInvoker();
             }
 
-            if (response.candidate && !gotstream) {
-                peer && peer.addICE({
+            if (response.candidate && !isGotRemoteStream) {
+                if (!peer) console.error('missed an ice', response.candidate);
+                else
+                    peer.addICE({
                         sdpMLineIndex: response.candidate.sdpMLineIndex,
                         candidate: JSON.parse(response.candidate.candidate)
                     });
             }
+
+            if (response.left) {
+                if (peer && peer.peer) {
+                    peer.peer.close();
+                    peer.peer = null;
+                }
+            }
         }
 
         var invokedOnce = false;
@@ -175,41 +148,46 @@ var rtclib = function (config) {
 
             invokedOnce = true;
 
-            inner.sdp = JSON.parse(inner.firstPart + inner.secondPart + inner.thirdPart);
             if (isofferer) peer.addAnswerSDP(inner.sdp);
             else initPeer(inner.sdp);
         }
     }
 
+    function leave() {
+        var length = sockets.length;
+        for (var i = 0; i < length; i++) {
+            var socket = sockets[i];
+            if (socket) {
+                socket.send({
+                    left: true,
+                    userToken: self.userToken
+                });
+                delete sockets[i];
+            }
+        }
+    }
+
+    window.onbeforeunload = function() {
+        leave();
+    };
+
+    window.onkeyup = function(e) {
+        if (e.keyCode == 116) leave();
+    };
+
     function startBroadcasting() {
-        defaultSocket && defaultSocket.send({
+        if (!isGotRemoteStream && defaultSocket) {
+            defaultSocket.send({
                 roomToken: self.roomToken,
                 roomName: self.roomName,
                 broadcaster: self.userToken
             });
+        }
         setTimeout(startBroadcasting, 3000);
     }
 
-    function onNewParticipant(channel) {
-        if (!channel || channels.indexOf(channel) != -1 || channel == self.userToken) return;
-        channels += channel + '--';
-
-        var new_channel = uniqueToken();
-        openSubSocket({
-                channel: new_channel,
-                closeSocket: true
-            });
-
-        defaultSocket.send({
-                participant: true,
-                userToken: self.userToken,
-                joinUser: channel,
-                channel: new_channel
-            });
-    }
-
     function uniqueToken() {
-        var s4 = function () {
+        var s4 = function() {
             return Math.floor(Math.random() * 0x10000).toString(16);
         };
         return s4() + s4() + "-" + s4() + "-" + s4() + "-" + s4() + "-" + s4() + s4() + s4();
@@ -217,7 +195,7 @@ var rtclib = function (config) {
 
     openDefaultSocket();
     return {
-        createRoom: function (_config) {
+        createRoom: function(_config) {
             self.roomName = _config.roomName || 'Anonymous';
             self.roomToken = uniqueToken();
 
@@ -225,19 +203,19 @@ var rtclib = function (config) {
             isGetNewRoom = false;
             startBroadcasting();
         },
-        joinRoom: function (_config) {
+        joinRoom: function(_config) {
             self.roomToken = _config.roomToken;
             isGetNewRoom = false;
 
             openSubSocket({
-                    channel: self.userToken
-                });
+                channel: self.userToken
+            });
 
             defaultSocket.send({
-                    participant: true,
-                    userToken: self.userToken,
-                    joinUser: _config.joinUser
-                });
+                participant: true,
+                userToken: self.userToken,
+                joinUser: _config.joinUser
+            });
         }
     };
 };
diff --git a/socket.io/ui.js b/socket.io/ui.js
index 7ff9ea54..b1361777 100644
--- a/socket.io/ui.js
+++ b/socket.io/ui.js
@@ -1,28 +1,51 @@
-// MIT License: https://webrtc-experiment.appspot.com/licence/
-// Use Socket.io over Node.js for signaling: https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socketio-over-nodejs
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socket.io
 
 var config = {
-    openSocket: function (config) {
-        var socket = io.connect('https://pubsub.pubnub.com/webrtc-socketio', {
-            publish_key: 'pub-f986077a-73bd-4c28-8e50-2e44076a84e0',
-            subscribe_key: 'sub-b8f4c07a-352e-11e2-bb9d-c7df1d04ae4a',
-            channel: config.channel || 'webrtc-socketio',
-            ssl: true
+    openSocket: function(config) {
+        var SIGNALING_SERVER = 'https://www.webrtc-experiment.com:8553/',
+            defaultChannel = location.hash.substr(1) || 'socketio-one-to-one-chat';
+
+        var channel = config.channel || defaultChannel;
+        var sender = Math.round(Math.random() * 999999999) + 999999999;
+
+        io.connect(SIGNALING_SERVER).emit('new-channel', {
+            channel: channel,
+            sender: sender
+        });
+
+        var socket = io.connect(SIGNALING_SERVER + channel);
+        socket.channel = channel;
+        socket.on('connect', function() {
+            if (config.callback) config.callback(socket);
         });
-        if (config.onopen) socket.on('connect', config.onopen);
+
+        socket.send = function(message) {
+            socket.emit('message', {
+                sender: sender,
+                data: message
+            });
+        };
+
         socket.on('message', config.onmessage);
-        return socket;
     },
-    onRemoteStream: function (media) {
+    onRemoteStream: function(media) {
         var video = media.video;
+
         video.setAttribute('controls', true);
+        video.setAttribute('id', media.stream.id);
 
         participants.insertBefore(video, participants.firstChild);
 
         video.play();
         rotateVideo(video);
     },
-    onRoomFound: function (room) {
+    onRemoteStreamEnded: function(stream) {
+        var video = document.getElementById(stream.id);
+        if (video) video.parentNode.removeChild(video);
+    },
+    onRoomFound: function(room) {
         var alreadyExist = document.getElementById(room.broadcaster);
         if (alreadyExist) return;
 
@@ -31,13 +54,13 @@ var config = {
         var tr = document.createElement('tr');
         tr.setAttribute('id', room.broadcaster);
         tr.innerHTML = '<td>' + room.roomName + '</td>' +
-            '<td><button class="join" id="' + room.roomToken + '">Join Room</button></td>';
+            '<td><button class="join" id="' + room.roomToken + '">Join</button></td>';
         roomsList.insertBefore(tr, roomsList.firstChild);
 
-        tr.onclick = function () {
+        tr.onclick = function() {
             var tr = this;
-            captureUserMedia(function () {
-                rtc.joinRoom({
+            captureUserMedia(function() {
+                rtcLib.joinRoom({
                     roomToken: tr.querySelector('.join').id,
                     joinUser: tr.id
                 });
@@ -48,9 +71,9 @@ var config = {
 };
 
 function createButtonClickHandler() {
-    captureUserMedia(function () {
-        rtc.createRoom({
-            roomName: (document.getElementById('room-name') || {}).value || 'Anonymous'
+    captureUserMedia(function() {
+        rtcLib.createRoom({
+            roomName: (document.getElementById('conference-name') || { }).value || 'Anonymous'
         });
     });
     hideUnnecessaryStuff();
@@ -60,27 +83,26 @@ function captureUserMedia(callback) {
     var video = document.createElement('video');
     video.setAttribute('autoplay', true);
     video.setAttribute('controls', true);
-
     participants.insertBefore(video, participants.firstChild);
 
     getUserMedia({
         video: video,
-        onsuccess: function (stream) {
+        onsuccess: function(stream) {
             config.attachStream = stream;
             callback && callback();
 
-            rotateVideo(video);
             video.setAttribute('muted', true);
+            rotateVideo(video);
         },
-        onerror: function (error) {
+        onerror: function() {
             alert('unable to get access to your webcam');
             callback && callback();
         }
     });
 }
 
-/* on page load: get public rooms */
-var rtc = rtclib(config);
+// You can use! window.onload = function() {}
+var rtcLib = RTCLib(config);
 
 /* UI specific */
 var participants = document.getElementById("participants") || document.body;
@@ -99,13 +121,14 @@ function hideUnnecessaryStuff() {
 
 function rotateVideo(video) {
     video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(0deg)';
-    setTimeout(function () {
+    setTimeout(function() {
         video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(360deg)';
     }, 1000);
 }
 
-(function () {
+(function() {
     var uniqueToken = document.getElementById('unique-token');
-    if (uniqueToken) if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
-    else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace(/\./g, '-');
-})();
\ No newline at end of file
+    if (uniqueToken)
+        if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
+        else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace( /\./g , '-');
+})();
diff --git a/socketio-over-nodejs/README.md b/socketio-over-nodejs/README.md
index c1f15397..0274d5d2 100644
--- a/socketio-over-nodejs/README.md
+++ b/socketio-over-nodejs/README.md
@@ -12,7 +12,7 @@ This experiment is using **socket.io over node.js** for signaling. Follow these
 
 `http://localhost:8888/` will be auto-opened.
 
-----
+=
 
 #### If you want to deploy your application
 
@@ -22,7 +22,7 @@ This experiment is using **socket.io over node.js** for signaling. Follow these
 
 and you're done!
 
-----
+=
 
 #### How to use?
 
@@ -59,7 +59,7 @@ connection.openSignalingChannel = function(config) {
 
 `io.connect(URL).emit('new-channel')` starts a new namespace that is used privately or publicly to transmit appropriate stuff e.g. room-details, participation-requests, SDP, ICE, etc.
 
-----
+=
 
 #### Presence Detection
 
@@ -81,7 +81,7 @@ testChannelPresence('default-channel');
 ```
 
 
-----
+=
 
 #### if fails to run....
 
@@ -102,19 +102,19 @@ childProcess.spawn(chromeURL, ['-incognito', openURL]);
 
 Actually, these lines are auto-opening chrome/firefox instances.
 
-----
+=
 
 #### What is `cd folder-location`?
 
 Using this command; you can open project's directory (i.e. folder).
 
-----
+=
 
 #### What is `node signaler`?
 
 This command runs the `signaler.js` file. That file handles socket.io server side stuff.
 
-----
+=
 
 #### What is `jitsu deploy`?
 
@@ -126,7 +126,7 @@ http://username.jit.su/
 
 See the demo URL: http://webrtc-signaling.jit.su/
 
-----
+=
 
 #### Note
 
@@ -144,7 +144,7 @@ It is strongly recommended to use absolute URL including port number:
 var SIGNALING_SERVER = 'http://domain.com:8888/';
 ```
 
-----
+=
 
 #### Are you beginner or totally novice?
 
@@ -153,14 +153,20 @@ var SIGNALING_SERVER = 'http://domain.com:8888/';
 3. You can use same command prompt to run any `node.js` file; also you can write `nodejitsu` commands in the same place e.g. `jitsu deploy` or `jitsu login` etc.
 4. Default port `8888` is used for this experiment. You can manually open this URL: `http://localhost:8888/`
 
-----
+=
 
 #### Signaling Concepts
 
 Interested to understand WebRTC Signaling Concepts? Read [this document](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/socketio-over-nodejs/Signaling-Concepts.md).
 
-----
+=
+
+#### SSL/TLS/HTTPS/Socket.io/Node.js ??
+
+https://github.com/muaz-khan/WebRTC-Experiment/issues/62
+
+=
 
 #### License
 
-[Socket.io over Node.js](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/socketio-over-nodejs) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[Socket.io over Node.js](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/socketio-over-nodejs) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/socketio-over-nodejs/Signaling-Concepts.md b/socketio-over-nodejs/Signaling-Concepts.md
index 9a12c436..fbc5a6ee 100644
--- a/socketio-over-nodejs/Signaling-Concepts.md
+++ b/socketio-over-nodejs/Signaling-Concepts.md
@@ -1,6 +1,8 @@
 #### WebRTC Experiments Signaling Concepts
 
-This document explains inner-parts of the signaling methods used in [WebRTC Experiments](https://webrtc-experiment.appspot.com/).
+This document explains inner-parts of the signaling methods used in [WebRTC Experiments](https://www.webrtc-experiment.com/).
+
+=
 
 #### Dynamic Channels
 
@@ -13,6 +15,8 @@ Signaling method must have following features:
 
 Out of `multiplexing` requirement; simple WebSocket implementation can't be used as signaling method in any WebRTC Experiment.
 
+=
+
 #### A sample socket.io implementation
 
 ```javascript
@@ -33,6 +37,8 @@ Node.js server is expected to fire two events:
 
 Node.js must catch messages passed from client-side and broadcast/transmit that message over all other connected sockets.
 
+=
+
 #### A full-fledge socket.io implementation / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/socketio-over-nodejs)
 
 Node.js server:
@@ -68,6 +74,8 @@ function onNewNamespace(channel, sender) {
 }
 ```
 
+=
+
 Client side:
 
 ```javascript
@@ -103,25 +111,33 @@ All WebRTC Experiments separated signaling portion; so you can define a single m
 
 `openSignalingChannel` or `openSocket` are kind of those "custom" reusable methods.
 
+=
+
 #### What about Firebase?
 
 Firebase is an io-based service stores data in JSON format; considered most suitable solution for WebRTC signaling. Its APIs are easier to use and understand.
 
+=
+
 #### What about PubNub or Pusher?
 
 PubNub or Pusher provides APIs for realtime connection. PubNub uses wider methods and techniques.
 
+=
+
 #### Expectations
 
 All WebRTC Experiments expects that signaling channels must be able to send and receive messages over unique channels.
 
+=
+
 #### Links
 
 1. Demo: http://webrtc-signaling.jit.su/
 2. Source code: https://github.com/muaz-khan/WebRTC-Experiment/blob/master/socketio-over-nodejs
 
-----
+=
 
 #### License
 
-[WebRTC Experiments](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC Experiments](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/socketio-over-nodejs/webrtc-signaling/README.md b/socketio-over-nodejs/webrtc-signaling/README.md
index c360f684..d23acf8f 100644
--- a/socketio-over-nodejs/webrtc-signaling/README.md
+++ b/socketio-over-nodejs/webrtc-signaling/README.md
@@ -2,7 +2,7 @@
 
 This directory contains only one file: [signaler.js](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/socketio-over-nodejs/webrtc-signaling/signaler.js) / socket.io main script
 
-----
+=
 
 #### [signaler.js](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/socketio-over-nodejs/webrtc-signaling/signaler.js)
 
@@ -41,8 +41,8 @@ function onNewNamespace(channel, sender) {
 }
 ```
 
-----
+=
 
 #### License
 
-[WebRTC Experiments](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC Experiments](https://github.com/muaz-khan/WebRTC-Experiment) are released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/socketio-over-nodejs/webrtc-signaling/static/RTCMultiConnection-v1.4/index.html b/socketio-over-nodejs/webrtc-signaling/static/RTCMultiConnection-v1.4/index.html
index 8d56a33d..ead9741d 100644
--- a/socketio-over-nodejs/webrtc-signaling/static/RTCMultiConnection-v1.4/index.html
+++ b/socketio-over-nodejs/webrtc-signaling/static/RTCMultiConnection-v1.4/index.html
@@ -1,5 +1,5 @@
 <script src="/socketio.js"></script>
-<script src="https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"></script>
 
 <button id="init">Open NEW Session</button>
 
diff --git a/socketio-over-nodejs/webrtc-signaling/static/RTCMultiConnection/index.html b/socketio-over-nodejs/webrtc-signaling/static/RTCMultiConnection/index.html
index d115470a..d13bc6a9 100644
--- a/socketio-over-nodejs/webrtc-signaling/static/RTCMultiConnection/index.html
+++ b/socketio-over-nodejs/webrtc-signaling/static/RTCMultiConnection/index.html
@@ -173,7 +173,7 @@
             .eject {
                 background-position: center center !important;
                 background-repeat: no-repeat !important;
-                background-image: url('https://webrtc-experiment.appspot.com/images/eject.png') !important;
+                background-image: url('https://www.webrtc-experiment.com/images/eject.png') !important;
                 width: 30px;
                 height: 30px;
                 position: absolute;
@@ -330,7 +330,7 @@ <h2>Remote videos container</h2>
             </script>
         </article>
         <footer>
-            <p>	<a href="https://webrtc-experiment.appspot.com/" target="_blank">https://webrtc-experiment.appspot.com/</a>
+            <p>	<a href="https://www.webrtc-experiment.com/" target="_blank">https://www.webrtc-experiment.com/</a>
 
             </p>
         </footer>
diff --git a/socketio-over-nodejs/webrtc-signaling/static/text-chat.html b/socketio-over-nodejs/webrtc-signaling/static/text-chat.html
index e506056a..06551142 100644
--- a/socketio-over-nodejs/webrtc-signaling/static/text-chat.html
+++ b/socketio-over-nodejs/webrtc-signaling/static/text-chat.html
@@ -122,7 +122,7 @@ <h1 style="text-align: center;display: block;border-bottom: 1px solid #CCCCCC;ma
         input.focus();
     }
 </script>
-<footer style="font-size: .8em; text-align: center;"> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> »
+<footer style="font-size: .8em; text-align: center;"> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> »
     <a
     href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a> »	<a href="/">Video Conferencing : HOME</a>
 
diff --git a/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/RTCPeerConnection-v1.5.js b/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/RTCPeerConnection-v1.5.js
index c4ca5796..41f0ba82 100644
--- a/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/RTCPeerConnection-v1.5.js
+++ b/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/RTCPeerConnection-v1.5.js
@@ -1,11 +1,10 @@
-/*
-     2013, @muazkh - github.com/muaz-khan
-     MIT License - https://webrtc-experiment.appspot.com/licence/
-     Documentation - https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCPeerConnection
-*/
-
-window.moz = !! navigator.mozGetUserMedia;
-var RTCPeerConnection = function (options) {
+// 2013, @muazkh - github.com/muaz-khan
+// MIT License - https://webrtc-experiment.appspot.com/licence/
+// Documentation - https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCPeerConnection
+
+window.moz = !!navigator.mozGetUserMedia;
+
+function RTCPeerConnection(options) {
     var w = window,
         PeerConnection = w.mozRTCPeerConnection || w.webkitRTCPeerConnection,
         SessionDescription = w.mozRTCSessionDescription || w.RTCSessionDescription,
@@ -25,7 +24,7 @@ var RTCPeerConnection = function (options) {
     };
 
     if (!moz && !options.iceServers) {
-        if (parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2]) >= 28)
+        if (parseInt(navigator.userAgent.match( /Chrom(e|ium)\/([0-9]+)\./ )[2]) >= 28)
             TURN = {
                 url: 'turn:turn.bistri.com:80',
                 credential: 'homeo',
@@ -50,27 +49,41 @@ var RTCPeerConnection = function (options) {
             }];
     }
 
-    var peerConnection = new PeerConnection(iceServers, optional);
+    var peer = new PeerConnection(iceServers, optional);
 
     openOffererChannel();
 
-    peerConnection.onicecandidate = onicecandidate;
-    if (options.attachStream) peerConnection.addStream(options.attachStream);
-    peerConnection.onaddstream = onaddstream;
+    peer.onicecandidate = function(event) {
+        if (event.candidate)
+            options.onICE(event.candidate);
+    };
+
+    // attachStream = MediaStream;
+    if (options.attachStream) peer.addStream(options.attachStream);
 
-    function onicecandidate(event) {
-        if (!event.candidate || !peerConnection) return;
-        if (options.onICE) options.onICE(event.candidate);
+    // attachStreams[0] = audio-stream;
+    // attachStreams[1] = video-stream;
+    // attachStreams[2] = screen-capturing-stream;
+    if (options.attachStreams && options.attachStream.length) {
+        var streams = options.attachStreams;
+        for (var i = 0; i < streams.length; i++) {
+            peer.addStream(streams[i]);
+        }
     }
 
-    var remoteStreamEventFired = false;
+    peer.onaddstream = function(event) {
+        var remoteMediaStream = event.stream;
 
-    function onaddstream(event) {
-        info('------------onaddstream');
-        if (remoteStreamEventFired || !event || !options.onRemoteStream) return;
-        remoteStreamEventFired = true;
-        options.onRemoteStream(event.stream);
-    }
+        // onRemoteStreamEnded(MediaStream)
+        remoteMediaStream.onended = function() {
+            if (options.onRemoteStreamEnded) options.onRemoteStreamEnded(remoteMediaStream);
+        };
+
+        // onRemoteStream(MediaStream)
+        if (options.onRemoteStream) options.onRemoteStream(remoteMediaStream);
+
+        console.debug('on:add:stream', remoteMediaStream);
+    };
 
     var constraints = options.constraints || {
         optional: [],
@@ -80,182 +93,175 @@ var RTCPeerConnection = function (options) {
         }
     };
 
-    var chars = 'ABCDEFGHIJKLMNOPQRSTUVWXYZ'.split(''),
-        extractedChars = '';
-
-    function getChars() {
-        extractedChars += chars[parseInt(Math.random() * 40)] || '';
-        if (extractedChars.length < 40) getChars();
-
-        return extractedChars;
-    }
-
-    function getInteropSDP(sdp) {
-        // for audio-only streaming: multiple-crypto lines are not allowed
-        if (options.onAnswerSDP)
-            sdp = sdp.replace(/(a=crypto:0 AES_CM_128_HMAC_SHA1_32)(.*?)(\r\n)/g, '');
-
-
-        var inline = getChars() + '\r\n' + (extractedChars = '');
-        sdp = sdp.indexOf('a=crypto') == -1 ? sdp.replace(/c=IN/g,
-            'a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:' + inline +
-            'c=IN') : sdp;
-
-        sdp = setBandwidth(sdp);
-
-        if (options.offerSDP) {
-            info('\n--------offer sdp provided by offerer\n');
-            info(options.offerSDP.sdp);
-        }
-
-        info(options.onOfferSDP ? '\n--------offer\n' : '\n--------answer\n');
-        info('sdp: ' + sdp);
-
-        return sdp;
-    }
+    // onOfferSDP(RTCSessionDescription)
 
     function createOffer() {
         if (!options.onOfferSDP) return;
 
-        peerConnection.createOffer(function (sessionDescription) {
-            sessionDescription.sdp = getInteropSDP(sessionDescription.sdp);
-            peerConnection.setLocalDescription(sessionDescription);
+        peer.createOffer(function(sessionDescription) {
+            sessionDescription.sdp = serializeSdp(sessionDescription.sdp);
+            peer.setLocalDescription(sessionDescription);
             options.onOfferSDP(sessionDescription);
         }, null, constraints);
     }
 
+    // onAnswerSDP(RTCSessionDescription)
+
     function createAnswer() {
         if (!options.onAnswerSDP) return;
 
-        options.offerSDP = new SessionDescription(options.offerSDP);
-        peerConnection.setRemoteDescription(options.offerSDP);
-
-        peerConnection.createAnswer(function (sessionDescription) {
-            sessionDescription.sdp = getInteropSDP(sessionDescription.sdp);
-            peerConnection.setLocalDescription(sessionDescription);
+        peer.setRemoteDescription(new SessionDescription(options.offerSDP));
+        peer.createAnswer(function(sessionDescription) {
+            sessionDescription.sdp = serializeSdp(sessionDescription.sdp);
+            peer.setLocalDescription(sessionDescription);
             options.onAnswerSDP(sessionDescription);
         }, null, constraints);
     }
 
-    function setBandwidth(sdp) {
-        // Firefox has no support of "b=AS"
-        if (moz) return sdp;
+    // if Mozilla Firefox & DataChannel; offer/answer will be created later
+    if ((options.onChannelMessage && !moz) || !options.onChannelMessage) {
+        createOffer();
+        createAnswer();
+    }
 
+
+    // DataChannel Bandwidth
+
+    function setBandwidth(sdp) {
         // remove existing bandwidth lines
-        sdp = sdp.replace(/b=AS([^\r\n]+\r\n)/g, '');
+        sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , '');
+        sdp = sdp.replace( /a=mid:data\r\n/g , 'a=mid:data\r\nb=AS:1638400\r\n');
+
+        return sdp;
+    }
+
+    // old: FF<>Chrome interoperability management
+
+    function getInteropSDP(sdp) {
+        var chars = 'ABCDEFGHIJKLMNOPQRSTUVWXYZ'.split(''),
+            extractedChars = '';
+
+        function getChars() {
+            extractedChars += chars[parseInt(Math.random() * 40)] || '';
+            if (extractedChars.length < 40)
+                getChars();
+
+            return extractedChars;
+        }
+
+        // for audio-only streaming: multiple-crypto lines are not allowed
+        if (options.onAnswerSDP)
+            sdp = sdp.replace( /(a=crypto:0 AES_CM_128_HMAC_SHA1_32)(.*?)(\r\n)/g , '');
 
-        sdp = sdp.replace(/a=mid:audio\r\n/g, 'a=mid:audio\r\nb=AS:50\r\n');
-        sdp = sdp.replace(/a=mid:video\r\n/g, 'a=mid:video\r\nb=AS:256\r\n');
-        sdp = sdp.replace(/a=mid:data\r\n/g, 'a=mid:data\r\nb=AS:1638400\r\n');
+        var inline = getChars() + '\r\n' + (extractedChars = '');
+        sdp = sdp.indexOf('a=crypto') == -1 ? sdp.replace( /c=IN/g ,
+            'a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:' + inline +
+                'c=IN') : sdp;
 
         return sdp;
     }
 
-    if ((options.onChannelMessage && !moz) || !options.onChannelMessage) {
-        createOffer();
-        createAnswer();
+    function serializeSdp(sdp) {
+        if (!moz) sdp = setBandwidth(sdp);
+        sdp = getInteropSDP(sdp);
+        console.debug(sdp);
+        return sdp;
     }
 
+    // DataChannel management
     var channel;
 
     function openOffererChannel() {
-        if (!options.onChannelMessage || (moz && !options.onOfferSDP)) return;
+        if (!options.onChannelMessage || (moz && !options.onOfferSDP))
+            return;
 
         _openOffererChannel();
 
-        if (moz && !options.attachStream) {
+        if (moz) {
             navigator.mozGetUserMedia({
-                audio: true,
-                fake: true
-            }, function (stream) {
-                peerConnection.addStream(stream);
-                createOffer();
-            }, useless);
+                    audio: true,
+                    fake: true
+                }, function(stream) {
+                    peer.addStream(stream);
+                    createOffer();
+                }, useless);
         }
     }
 
     function _openOffererChannel() {
-        channel = peerConnection.createDataChannel(
-            options.channel || 'RTCDataChannel',
-            moz ? {} : {
-                reliable: false
-            });
+        channel = peer.createDataChannel(options.channel || 'RTCDataChannel', moz ? { } : {
+            reliable: false
+        });
 
-        if (moz) channel.binaryType = 'blob';
+        if (moz)
+            channel.binaryType = 'blob';
         setChannelEvents();
     }
 
     function setChannelEvents() {
-        channel.onmessage = function (event) {
+        channel.onmessage = function(event) {
             if (options.onChannelMessage) options.onChannelMessage(event);
         };
 
-        channel.onopen = function () {
+        channel.onopen = function() {
             if (options.onChannelOpened) options.onChannelOpened(channel);
         };
-        channel.onclose = function (event) {
+        channel.onclose = function(event) {
             if (options.onChannelClosed) options.onChannelClosed(event);
 
             console.warn('WebRTC DataChannel closed', event);
         };
-        channel.onerror = function (event) {
-            console.error(event);
+        channel.onerror = function(event) {
             if (options.onChannelError) options.onChannelError(event);
         };
     }
 
-    if (options.onAnswerSDP && moz) openAnswererChannel();
+    if (options.onAnswerSDP && moz && options.onChannelMessage)
+        openAnswererChannel();
 
     function openAnswererChannel() {
-        peerConnection.ondatachannel = function (_channel) {
-            channel = _channel;
+        peer.ondatachannel = function(event) {
+            channel = event.channel;
             channel.binaryType = 'blob';
             setChannelEvents();
         };
 
-        if (moz && !options.attachStream) {
+        if (moz) {
             navigator.mozGetUserMedia({
-                audio: true,
-                fake: true
-            }, function (stream) {
-                peerConnection.addStream(stream);
-                createAnswer();
-            }, useless);
+                    audio: true,
+                    fake: true
+                }, function(stream) {
+                    peer.addStream(stream);
+                    createAnswer();
+                }, useless);
         }
     }
 
-    function useless() {}
-
-    function info(information) {
-        console.log(information);
+    function useless() {
     }
 
     return {
-        addAnswerSDP: function (sdp) {
-            info('--------adding answer sdp:');
-            info(sdp.sdp);
-
-            sdp = new SessionDescription(sdp);
-            peerConnection.setRemoteDescription(sdp);
+        addAnswerSDP: function(sdp) {
+            peer.setRemoteDescription(new SessionDescription(sdp));
         },
-        addICE: function (candidate) {
-            info(candidate.candidate);
-            peerConnection.addIceCandidate(new IceCandidate({
+        addICE: function(candidate) {
+            peer.addIceCandidate(new IceCandidate({
                 sdpMLineIndex: candidate.sdpMLineIndex,
                 candidate: candidate.candidate
             }));
         },
 
-        peer: peerConnection,
+        peer: peer,
         channel: channel,
-        sendData: function (message) {
+        sendData: function(message) {
             channel && channel.send(message);
         }
     };
-};
+}
 
+// getUserMedia
 var video_constraints = {
-    mandatory: {},
+    mandatory: { },
     optional: []
 };
 
@@ -264,11 +270,11 @@ function getUserMedia(options) {
         media;
     n.getMedia = n.webkitGetUserMedia || n.mozGetUserMedia;
     n.getMedia(options.constraints || {
-        audio: true,
-        video: video_constraints
-    }, streaming, options.onerror || function (e) {
-        console.error(e);
-    });
+            audio: true,
+            video: video_constraints
+        }, streaming, options.onerror || function(e) {
+            console.error(e);
+        });
 
     function streaming(stream) {
         var video = options.video;
diff --git a/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/conference-ui.js b/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/conference-ui.js
index b32c52f4..bce72877 100644
--- a/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/conference-ui.js
+++ b/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/conference-ui.js
@@ -1,41 +1,51 @@
-/* MIT License: https://webrtc-experiment.appspot.com/licence/ */
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing
 
-config = {
-    openSocket    : function (config) {
-        var SIGNALING_SERVER = '/';
-        var channel = config.channel || location.hash.replace('#', '') || 'video-conferencing';
-        var sender = Math.round(Math.random() * 60535) + 5000;
+var config = {
+    openSocket: function(config) {
+        var SIGNALING_SERVER = '/',
+            defaultChannel = location.hash.substr(1) || 'video-conferencing-hangout';
+
+        var channel = config.channel || defaultChannel;
+        var sender = Math.round(Math.random() * 999999999) + 999999999;
 
         io.connect(SIGNALING_SERVER).emit('new-channel', {
             channel: channel,
-            sender : sender
+            sender: sender
         });
 
         var socket = io.connect(SIGNALING_SERVER + channel);
         socket.channel = channel;
-        socket.on('connect', function () {
+        socket.on('connect', function() {
             if (config.callback) config.callback(socket);
         });
 
-        socket.send = function (message) {
+        socket.send = function(message) {
             socket.emit('message', {
                 sender: sender,
-                data  : message
+                data: message
             });
         };
 
         socket.on('message', config.onmessage);
     },
-    onRemoteStream: function (media) {
+    onRemoteStream: function(media) {
         var video = media.video;
+
         video.setAttribute('controls', true);
+        video.setAttribute('id', media.stream.id);
 
         participants.insertBefore(video, participants.firstChild);
 
         video.play();
         rotateVideo(video);
     },
-    onRoomFound   : function (room) {
+    onRemoteStreamEnded: function(stream) {
+        var video = document.getElementById(stream.id);
+        if (video) video.parentNode.removeChild(video);
+    },
+    onRoomFound: function(room) {
         var alreadyExist = document.getElementById(room.broadcaster);
         if (alreadyExist) return;
 
@@ -44,15 +54,15 @@ config = {
         var tr = document.createElement('tr');
         tr.setAttribute('id', room.broadcaster);
         tr.innerHTML = '<td>' + room.roomName + '</td>' +
-            '<td><button class="join" id="' + room.roomToken + '">Join Room</button></td>';
+            '<td><button class="join" id="' + room.roomToken + '">Join Conference</button></td>';
         roomsList.insertBefore(tr, roomsList.firstChild);
 
-        tr.onclick = function () {
+        tr.onclick = function() {
             var tr = this;
-            captureUserMedia(function () {
+            captureUserMedia(function() {
                 conferenceUI.joinRoom({
                     roomToken: tr.querySelector('.join').id,
-                    joinUser : tr.id
+                    joinUser: tr.id
                 });
             });
             hideUnnecessaryStuff();
@@ -61,7 +71,7 @@ config = {
 };
 
 function createButtonClickHandler() {
-    captureUserMedia(function () {
+    captureUserMedia(function() {
         conferenceUI.createRoom({
             roomName: (document.getElementById('conference-name') || { }).value || 'Anonymous'
         });
@@ -76,28 +86,28 @@ function captureUserMedia(callback) {
     participants.insertBefore(video, participants.firstChild);
 
     getUserMedia({
-        video    : video,
-        onsuccess: function (stream) {
+        video: video,
+        onsuccess: function(stream) {
             config.attachStream = stream;
             callback && callback();
 
             video.setAttribute('muted', true);
             rotateVideo(video);
         },
-        onerror  : function () {
+        onerror: function() {
             alert('unable to get access to your webcam');
             callback && callback();
         }
     });
 }
 
-/* on page load: get public rooms */
-conferenceUI = conference(config);
+// You can use! window.onload = function() {}
+var conferenceUI = conference(config);
 
 /* UI specific */
-participants = document.getElementById("participants") || document.body;
-startConferencing = document.getElementById('start-conferencing');
-roomsList = document.getElementById('rooms-list');
+var participants = document.getElementById("participants") || document.body;
+var startConferencing = document.getElementById('start-conferencing');
+var roomsList = document.getElementById('rooms-list');
 
 if (startConferencing) startConferencing.onclick = createButtonClickHandler;
 
@@ -111,17 +121,14 @@ function hideUnnecessaryStuff() {
 
 function rotateVideo(video) {
     video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(0deg)';
-    setTimeout(function () {
+    setTimeout(function() {
         video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(360deg)';
     }, 1000);
 }
 
-(function () {
-    uniqueToken = document.getElementById('unique-token');
-    if (uniqueToken) if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">You can share this private link with your friends.</a></h2>';
-    else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = (function () {
-            return "#private-" + ("" + 1e10).replace(/[018]/g, function (a) {
-                return (a ^ Math.random() * 16 >> a / 4).toString(16);
-            });
-        })();
-})();
\ No newline at end of file
+(function() {
+    var uniqueToken = document.getElementById('unique-token');
+    if (uniqueToken)
+        if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
+        else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace( /\./g , '-');
+})();
diff --git a/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/conference.js b/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/conference.js
index 9be610f2..8607a54d 100644
--- a/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/conference.js
+++ b/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/conference.js
@@ -1,18 +1,21 @@
-/* MIT License: https://webrtc-experiment.appspot.com/licence/ 
- It is recommended to use RTCMultiConnection.js for audio/video/screen sharing: <http://bit.ly/RTCMultiConnection-Documentation>
- */
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing
 
-var conference = function (config) {
-    var self = { userToken: uniqueToken() },
-        channels = '--',
+var conference = function(config) {
+    var self = {
+        userToken: uniqueToken()
+    };
+    var channels = '--',
         isbroadcaster,
         isGetNewRoom = true,
-        defaultSocket = {};
+        sockets = [],
+        defaultSocket = { };
 
     function openDefaultSocket() {
         defaultSocket = config.openSocket({
             onmessage: onDefaultSocketResponse,
-            callback : function (socket) {
+            callback: function(socket) {
                 defaultSocket = socket;
             }
         });
@@ -28,9 +31,8 @@ var conference = function (config) {
         if (response.userToken && response.joinUser == self.userToken && response.participant && channels.indexOf(response.userToken) == -1) {
             channels += response.userToken + '--';
             openSubSocket({
-                isofferer  : true,
-                channel    : response.channel || response.userToken,
-                closeSocket: true
+                isofferer: true,
+                channel: response.channel || response.userToken
             });
         }
     }
@@ -38,33 +40,38 @@ var conference = function (config) {
     function openSubSocket(_config) {
         if (!_config.channel) return;
         var socketConfig = {
-            channel  : _config.channel,
+            channel: _config.channel,
             onmessage: socketResponse,
-            callback : function (_socket) {
-                socket = _socket;
+            onopen: function() {
                 if (isofferer && !peer) initPeer();
+                sockets[sockets.length] = socket;
             }
         };
 
+        socketConfig.callback = function(_socket) {
+            socket = _socket;
+            this.onopen();
+        };
+
         var socket = config.openSocket(socketConfig),
             isofferer = _config.isofferer,
             gotstream,
             video = document.createElement('video'),
-            inner = {},
+            inner = { },
             peer;
 
         var peerConfig = {
-            attachStream  : config.attachStream,
-            onICE         : function (candidate) {
+            attachStream: config.attachStream,
+            onICE: function(candidate) {
                 socket.send({
                     userToken: self.userToken,
                     candidate: {
                         sdpMLineIndex: candidate.sdpMLineIndex,
-                        candidate    : JSON.stringify(candidate.candidate)
+                        candidate: JSON.stringify(candidate.candidate)
                     }
                 });
             },
-            onRemoteStream: function (stream) {
+            onRemoteStream: function(stream) {
                 if (!stream) return;
 
                 video[moz ? 'mozSrcObject' : 'src'] = moz ? stream : webkitURL.createObjectURL(stream);
@@ -72,6 +79,10 @@ var conference = function (config) {
 
                 _config.stream = stream;
                 onRemoteStreamStartsFlowing();
+            },
+            onRemoteStreamEnded: function(stream) {
+                if (config.onRemoteStreamEnded)
+                    config.onRemoteStreamEnded(stream);
             }
         };
 
@@ -90,77 +101,51 @@ var conference = function (config) {
             if (!(video.readyState <= HTMLMediaElement.HAVE_CURRENT_DATA || video.paused || video.currentTime <= 0)) {
                 gotstream = true;
 
-                config.onRemoteStream({
-                    video : video,
-                    stream: _config.stream
-                });
+                if (config.onRemoteStream)
+                    config.onRemoteStream({
+                        video: video,
+                        stream: _config.stream
+                    });
 
                 if (isbroadcaster && channels.split('--').length > 3) {
                     /* broadcasting newly connected participant for video-conferencing! */
                     defaultSocket.send({
                         newParticipant: socket.channel,
-                        userToken     : self.userToken
+                        userToken: self.userToken
                     });
                 }
 
-                /* closing subsocket here on the offerer side */
-                if (_config.closeSocket) socket = null;
-
             } else setTimeout(onRemoteStreamStartsFlowing, 50);
         }
 
         function sendsdp(sdp) {
-            sdp = JSON.stringify(sdp);
-            var part = parseInt(sdp.length / 3);
-
-            var firstPart = sdp.slice(0, part),
-                secondPart = sdp.slice(part, sdp.length - 1),
-                thirdPart = '';
-
-            if (sdp.length > part + part) {
-                secondPart = sdp.slice(part, part + part);
-                thirdPart = sdp.slice(part + part, sdp.length);
-            }
-
             socket.send({
                 userToken: self.userToken,
-                firstPart: firstPart
-            });
-
-            socket.send({
-                userToken : self.userToken,
-                secondPart: secondPart
-            });
-
-            socket.send({
-                userToken: self.userToken,
-                thirdPart: thirdPart
+                sdp: JSON.stringify(sdp)
             });
         }
 
         function socketResponse(response) {
             if (response.userToken == self.userToken) return;
-            if (response.firstPart || response.secondPart || response.thirdPart) {
-                if (response.firstPart) {
-                    inner.firstPart = response.firstPart;
-                    if (inner.secondPart && inner.thirdPart) selfInvoker();
-                }
-                if (response.secondPart) {
-                    inner.secondPart = response.secondPart;
-                    if (inner.firstPart && inner.thirdPart) selfInvoker();
-                }
-
-                if (response.thirdPart) {
-                    inner.thirdPart = response.thirdPart;
-                    if (inner.firstPart && inner.secondPart) selfInvoker();
-                }
+            if (response.sdp) {
+                inner.sdp = JSON.parse(response.sdp);
+                selfInvoker();
             }
 
             if (response.candidate && !gotstream) {
-                peer && peer.addICE({
-                    sdpMLineIndex: response.candidate.sdpMLineIndex,
-                    candidate    : JSON.parse(response.candidate.candidate)
-                });
+                if (!peer) console.error('missed an ice', response.candidate);
+                else
+                    peer.addICE({
+                        sdpMLineIndex: response.candidate.sdpMLineIndex,
+                        candidate: JSON.parse(response.candidate.candidate)
+                    });
+            }
+
+            if (response.left) {
+                if (peer && peer.peer) {
+                    peer.peer.close();
+                    peer.peer = null;
+                }
             }
         }
 
@@ -171,16 +156,37 @@ var conference = function (config) {
 
             invokedOnce = true;
 
-            inner.sdp = JSON.parse(inner.firstPart + inner.secondPart + inner.thirdPart);
             if (isofferer) peer.addAnswerSDP(inner.sdp);
             else initPeer(inner.sdp);
         }
     }
 
+    function leave() {
+        var length = sockets.length;
+        for (var i = 0; i < length; i++) {
+            var socket = sockets[i];
+            if (socket) {
+                socket.send({
+                    left: true,
+                    userToken: self.userToken
+                });
+                delete sockets[i];
+            }
+        }
+    }
+
+    window.onbeforeunload = function() {
+        leave();
+    };
+
+    window.onkeyup = function(e) {
+        if (e.keyCode == 116) leave();
+    };
+
     function startBroadcasting() {
         defaultSocket && defaultSocket.send({
-            roomToken  : self.roomToken,
-            roomName   : self.roomName,
+            roomToken: self.roomToken,
+            roomName: self.roomName,
             broadcaster: self.userToken
         });
         setTimeout(startBroadcasting, 3000);
@@ -192,20 +198,19 @@ var conference = function (config) {
 
         var new_channel = uniqueToken();
         openSubSocket({
-            channel    : new_channel,
-            closeSocket: true
+            channel: new_channel
         });
 
         defaultSocket.send({
             participant: true,
-            userToken  : self.userToken,
-            joinUser   : channel,
-            channel    : new_channel
+            userToken: self.userToken,
+            joinUser: channel,
+            channel: new_channel
         });
     }
 
     function uniqueToken() {
-        var s4 = function () {
+        var s4 = function() {
             return Math.floor(Math.random() * 0x10000).toString(16);
         };
         return s4() + s4() + "-" + s4() + "-" + s4() + "-" + s4() + "-" + s4() + s4() + s4();
@@ -213,7 +218,7 @@ var conference = function (config) {
 
     openDefaultSocket();
     return {
-        createRoom: function (_config) {
+        createRoom: function(_config) {
             self.roomName = _config.roomName || 'Anonymous';
             self.roomToken = uniqueToken();
 
@@ -221,7 +226,7 @@ var conference = function (config) {
             isGetNewRoom = false;
             startBroadcasting();
         },
-        joinRoom  : function (_config) {
+        joinRoom: function(_config) {
             self.roomToken = _config.roomToken;
             isGetNewRoom = false;
 
@@ -231,9 +236,9 @@ var conference = function (config) {
 
             defaultSocket.send({
                 participant: true,
-                userToken  : self.userToken,
-                joinUser   : _config.joinUser
+                userToken: self.userToken,
+                joinUser: _config.joinUser
             });
         }
     };
-};
\ No newline at end of file
+};
diff --git a/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/index.html b/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/index.html
index 85e99577..a112951a 100644
--- a/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/index.html
+++ b/socketio-over-nodejs/webrtc-signaling/static/video-conferencing/index.html
@@ -145,7 +145,7 @@
     </head>
     
     <body>
-        <article> <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
+        <article> <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑
         WEBRTC EXPERIMENTS</a>
 
              <h1>
@@ -157,7 +157,7 @@ <h1>
 
             </p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <table class="visible">
                 <tr>
@@ -200,12 +200,12 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                 href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+        <script src="https://www.webrtc-experiment.com/common.js"></script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/text-chat/README.md b/text-chat/README.md
index fbd8a9d3..f2593fb7 100644
--- a/text-chat/README.md
+++ b/text-chat/README.md
@@ -1,4 +1,4 @@
-#### WebRTC Text Chat i.e. Data Sharing / [Demo](https://webrtc-experiment.appspot.com/text-chat/)
+#### WebRTC Text Chat i.e. Data Sharing / [Demo](https://www.webrtc-experiment.com/text-chat/)
 
 `DataConnection.js` library lets you:
 
@@ -6,15 +6,15 @@
 2. Share text message of any length
 3. Text data regardless of the size and type
 
-----
+=
 
 #### First Step: Link the library
 
 ```html
-<script src="https://webrtc-experiment.appspot.com/data-connection.js"></script>
+<script src="https://www.webrtc-experiment.com/data-connection.js"></script>
 ```
 
-----
+=
 
 #### Last Step: Start using it!
 
@@ -29,7 +29,7 @@ document.getElementById('setup-new-connection').onclick = function() {
 };
 ```
 
-----
+=
 
 #### Text Chat i.e. Text Sharing
 
@@ -43,7 +43,7 @@ You may want to share direct messages:
 connection.channels['user-id'].send('longest possible text message');
 ```
 
-----
+=
 
 #### Errors Handling
 
@@ -55,7 +55,7 @@ connection.onerror = function(event, userid) {}
 connection.onclose = function(event, userid) {}
 ```
 
-----
+=
 
 #### Custom user-ids?
 
@@ -63,7 +63,7 @@ connection.onclose = function(event, userid) {}
 connection.userid = 'username';
 ```
 
-----
+=
 
 #### Custom signaling channel?
 
@@ -102,7 +102,7 @@ connection.firebase = 'chat';
 
 Want to use XHR, WebSockets, SIP, XMPP, etc. for signaling? Read [this post](https://github.com/muaz-khan/WebRTC-Experiment/issues/56#issuecomment-20090650).
 
-----
+=
 
 #### Want to manually join rooms?
 
@@ -117,7 +117,7 @@ connection.onconnection = function(room) {
 
 `onconnection` is called for each new data connection; and `join` method allows you manually join previously created connections.
 
-----
+=
 
 #### If someone leaves...
 
@@ -129,11 +129,11 @@ connection.onuserleft = function(userid) {
 };
 ```
 
-----
+=
 
 #### Browser Support
 
-This [DataConnection.js](https://webrtc-experiment.appspot.com/data-connection.js) library is compatible to following web-browsers:
+This [DataConnection.js](https://www.webrtc-experiment.com/data-connection.js) library is compatible to following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -141,8 +141,8 @@ This [DataConnection.js](https://webrtc-experiment.appspot.com/data-connection.j
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[DataConnection.js](https://webrtc-experiment.appspot.com/data-connection.js) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[DataConnection.js](https://www.webrtc-experiment.com/data-connection.js) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/text-chat/index.html b/text-chat/index.html
index 9867965c..59f360e8 100644
--- a/text-chat/index.html
+++ b/text-chat/index.html
@@ -132,8 +132,8 @@
                 margin-left: 6em;
             }
         </style>
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/data-connection.js"> </script>
+        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="https://www.webrtc-experiment.com/data-connection.js"> </script>
     </head>
     <body>
         <h1>
@@ -145,7 +145,7 @@ <h1>
                                               Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span>
         </p>
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
         <section>
             <h2>Setup a new connection:</h2>
@@ -256,7 +256,7 @@ <h2 style="display: block; font-size: 1em; text-align: center;">
         <br />
         <br />
         <pre>
-&lt;script src="https://webrtc-experiment.appspot.com/data-connection.js"&gt;&lt;/script&gt;
+&lt;script src="https://www.webrtc-experiment.com/data-connection.js"&gt;&lt;/script&gt;
 </pre>
         <pre>
 var connection = new DataConnection();
@@ -293,8 +293,8 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             <button id="send-message" style="font-size: 1em;">Send Message</button>
         </section>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a> © <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span> » <a href="mailto:muazkh@gmail.com" target="_blank">Email</a> » <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a> » <a href="https://github.com/muaz-khan" target="_blank">Github</a></p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+        <script src="https://www.webrtc-experiment.com/common.js"> </script>
     </body>
 </html>
diff --git a/video-conferencing/README.md b/video-conferencing/README.md
index 37be9fbd..b8f24b94 100644
--- a/video-conferencing/README.md
+++ b/video-conferencing/README.md
@@ -1,6 +1,6 @@
-#### WebRTC Group video sharing / [Demo](https://webrtc-experiment.appspot.com/video-conferencing/)
+#### WebRTC Group video sharing / [Demo](https://www.webrtc-experiment.com/video-conferencing/)
 
-----
+=
 
 #### How `video conferencing` Works?
 
@@ -11,6 +11,8 @@ For 10 people sharing videos in a group:
 1. Creating 10 `unique` peer connections
 2. Opening 10 unique `sockets` to exchange SDP/ICE
 
+=
+
 For your information; in One-to-One video session; 4 RTP streams/ports get open:
 
 1. One RTP port for **outgoing video**
@@ -18,12 +20,16 @@ For your information; in One-to-One video session; 4 RTP streams/ports get open:
 3. One RTP port for **incoming video**
 4. One RTP port for **incoming audio**
 
+=
+
 So, for 10 peers sharing video in a group; `40 RTP` ports get open. Which causes:
 
 1. Blurry video experience
 2. Unclear voice
 3. Bandwidth issues / slow streaming
 
+=
+
 The best solution is to use a **middle media server** like **asterisk** or **kamailio** to broadcast your camera stream. 
 
 To overcome burden and to deliver HD stream over thousands of peers; we need a media server that should **broadcast** stream coming from room owner's side.
@@ -53,11 +59,11 @@ Media server should mix all video streams; and stream it over single RTP port.
 
 If 10 room participants are sending video streams; server should mix them to generate a single media stream; then send that stream over single **incoming** RTP port opened between server and **room initiator**.
 
-----
+=
 
 #### Browser Support
 
-This [WebRTC Video Conferencing](https://webrtc-experiment.appspot.com/video-conferencing/) experiment works fine on following web-browsers:
+This [WebRTC Video Conferencing](https://www.webrtc-experiment.com/video-conferencing/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -65,8 +71,8 @@ This [WebRTC Video Conferencing](https://webrtc-experiment.appspot.com/video-con
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[WebRTC Video Conferencing](https://webrtc-experiment.appspot.com/video-conferencing/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC Video Conferencing](https://www.webrtc-experiment.com/video-conferencing/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/video-conferencing/conference-ui.js b/video-conferencing/conference-ui.js
index 161647c5..b3ff657c 100644
--- a/video-conferencing/conference-ui.js
+++ b/video-conferencing/conference-ui.js
@@ -1,34 +1,51 @@
-/*
-     2013, @muazkh - github.com/muaz-khan
-     MIT License - https://webrtc-experiment.appspot.com/licence/
-     Documentation - https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection
-*/
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing
 
 var config = {
-    openSocket: function (config) {
-        var channel = config.channel || location.hash.replace('#', '') || 'video-conferencing';
-        var socket = new Firebase('https://webrtc-experiment.firebaseIO.com/' + channel);
+    openSocket: function(config) {
+        var SIGNALING_SERVER = 'https://www.webrtc-experiment.com:8553/',
+            defaultChannel = location.hash.substr(1) || 'video-conferencing-hangout';
+
+        var channel = config.channel || defaultChannel;
+        var sender = Math.round(Math.random() * 999999999) + 999999999;
+
+        io.connect(SIGNALING_SERVER).emit('new-channel', {
+            channel: channel,
+            sender: sender
+        });
+
+        var socket = io.connect(SIGNALING_SERVER + channel);
         socket.channel = channel;
-        socket.on('child_added', function (data) {
-            config.onmessage(data.val());
+        socket.on('connect', function() {
+            if (config.callback) config.callback(socket);
         });
-        socket.send = function (data) {
-            this.push(data);
-        }
-        config.onopen && setTimeout(config.onopen, 1);
-        socket.onDisconnect().remove();
-        return socket;
+
+        socket.send = function(message) {
+            socket.emit('message', {
+                sender: sender,
+                data: message
+            });
+        };
+
+        socket.on('message', config.onmessage);
     },
-    onRemoteStream: function (media) {
+    onRemoteStream: function(media) {
         var video = media.video;
+
         video.setAttribute('controls', true);
+        video.setAttribute('id', media.stream.id);
 
         participants.insertBefore(video, participants.firstChild);
 
         video.play();
         rotateVideo(video);
     },
-    onRoomFound: function (room) {
+    onRemoteStreamEnded: function(stream) {
+        var video = document.getElementById(stream.id);
+        if (video) video.parentNode.removeChild(video);
+    },
+    onRoomFound: function(room) {
         var alreadyExist = document.getElementById(room.broadcaster);
         if (alreadyExist) return;
 
@@ -40,9 +57,9 @@ var config = {
             '<td><button class="join" id="' + room.roomToken + '">Join Conference</button></td>';
         roomsList.insertBefore(tr, roomsList.firstChild);
 
-        tr.onclick = function () {
+        tr.onclick = function() {
             var tr = this;
-            captureUserMedia(function () {
+            captureUserMedia(function() {
                 conferenceUI.joinRoom({
                     roomToken: tr.querySelector('.join').id,
                     joinUser: tr.id
@@ -54,7 +71,7 @@ var config = {
 };
 
 function createButtonClickHandler() {
-    captureUserMedia(function () {
+    captureUserMedia(function() {
         conferenceUI.createRoom({
             roomName: (document.getElementById('conference-name') || { }).value || 'Anonymous'
         });
@@ -70,21 +87,21 @@ function captureUserMedia(callback) {
 
     getUserMedia({
         video: video,
-        onsuccess: function (stream) {
+        onsuccess: function(stream) {
             config.attachStream = stream;
             callback && callback();
 
             video.setAttribute('muted', true);
             rotateVideo(video);
         },
-        onerror: function () {
+        onerror: function() {
             alert('unable to get access to your webcam');
             callback && callback();
         }
     });
 }
 
-/* on page load: get public rooms */
+// You can use! window.onload = function() {}
 var conferenceUI = conference(config);
 
 /* UI specific */
@@ -104,13 +121,14 @@ function hideUnnecessaryStuff() {
 
 function rotateVideo(video) {
     video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(0deg)';
-    setTimeout(function () {
+    setTimeout(function() {
         video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(360deg)';
     }, 1000);
 }
 
-(function () {
+(function() {
     var uniqueToken = document.getElementById('unique-token');
-    if (uniqueToken) if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
-    else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace(/\./g, '-');
-})();
\ No newline at end of file
+    if (uniqueToken)
+        if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '<h2 style="text-align:center;"><a href="' + location.href + '" target="_blank">Share this link</a></h2>';
+        else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace( /\./g , '-');
+})();
diff --git a/video-conferencing/conference.js b/video-conferencing/conference.js
index 2ecf9bb3..8607a54d 100644
--- a/video-conferencing/conference.js
+++ b/video-conferencing/conference.js
@@ -1,21 +1,24 @@
-/* MIT License: https://webrtc-experiment.appspot.com/licence/ */
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing
 
-var conference = function (config) {
+var conference = function(config) {
     var self = {
         userToken: uniqueToken()
-    },
-        channels = '--',
+    };
+    var channels = '--',
         isbroadcaster,
         isGetNewRoom = true,
-        defaultSocket = {};
+        sockets = [],
+        defaultSocket = { };
 
     function openDefaultSocket() {
         defaultSocket = config.openSocket({
-                onmessage: onDefaultSocketResponse,
-                callback: function (socket) {
-                    defaultSocket = socket;
-                }
-            });
+            onmessage: onDefaultSocketResponse,
+            callback: function(socket) {
+                defaultSocket = socket;
+            }
+        });
     }
 
     function onDefaultSocketResponse(response) {
@@ -28,10 +31,9 @@ var conference = function (config) {
         if (response.userToken && response.joinUser == self.userToken && response.participant && channels.indexOf(response.userToken) == -1) {
             channels += response.userToken + '--';
             openSubSocket({
-                    isofferer: true,
-                    channel: response.channel || response.userToken,
-                    closeSocket: true
-                });
+                isofferer: true,
+                channel: response.channel || response.userToken
+            });
         }
     }
 
@@ -40,12 +42,13 @@ var conference = function (config) {
         var socketConfig = {
             channel: _config.channel,
             onmessage: socketResponse,
-            onopen: function () {
+            onopen: function() {
                 if (isofferer && !peer) initPeer();
+                sockets[sockets.length] = socket;
             }
         };
 
-        socketConfig.callback = function (_socket) {
+        socketConfig.callback = function(_socket) {
             socket = _socket;
             this.onopen();
         };
@@ -54,21 +57,21 @@ var conference = function (config) {
             isofferer = _config.isofferer,
             gotstream,
             video = document.createElement('video'),
-            inner = {},
+            inner = { },
             peer;
 
         var peerConfig = {
             attachStream: config.attachStream,
-            onICE: function (candidate) {
+            onICE: function(candidate) {
                 socket.send({
-                        userToken: self.userToken,
-                        candidate: {
-                            sdpMLineIndex: candidate.sdpMLineIndex,
-                            candidate: JSON.stringify(candidate.candidate)
-                        }
-                    });
+                    userToken: self.userToken,
+                    candidate: {
+                        sdpMLineIndex: candidate.sdpMLineIndex,
+                        candidate: JSON.stringify(candidate.candidate)
+                    }
+                });
             },
-            onRemoteStream: function (stream) {
+            onRemoteStream: function(stream) {
                 if (!stream) return;
 
                 video[moz ? 'mozSrcObject' : 'src'] = moz ? stream : webkitURL.createObjectURL(stream);
@@ -76,6 +79,10 @@ var conference = function (config) {
 
                 _config.stream = stream;
                 onRemoteStreamStartsFlowing();
+            },
+            onRemoteStreamEnded: function(stream) {
+                if (config.onRemoteStreamEnded)
+                    config.onRemoteStreamEnded(stream);
             }
         };
 
@@ -94,7 +101,8 @@ var conference = function (config) {
             if (!(video.readyState <= HTMLMediaElement.HAVE_CURRENT_DATA || video.paused || video.currentTime <= 0)) {
                 gotstream = true;
 
-                config.onRemoteStream({
+                if (config.onRemoteStream)
+                    config.onRemoteStream({
                         video: video,
                         stream: _config.stream
                     });
@@ -102,70 +110,43 @@ var conference = function (config) {
                 if (isbroadcaster && channels.split('--').length > 3) {
                     /* broadcasting newly connected participant for video-conferencing! */
                     defaultSocket.send({
-                            newParticipant: socket.channel,
-                            userToken: self.userToken
-                        });
+                        newParticipant: socket.channel,
+                        userToken: self.userToken
+                    });
                 }
 
-                /* closing subsocket here on the offerer side */
-                if (_config.closeSocket) socket = null;
-
             } else setTimeout(onRemoteStreamStartsFlowing, 50);
         }
 
         function sendsdp(sdp) {
-            sdp = JSON.stringify(sdp);
-            var part = parseInt(sdp.length / 3);
-
-            var firstPart = sdp.slice(0, part),
-                secondPart = sdp.slice(part, sdp.length - 1),
-                thirdPart = '';
-
-            if (sdp.length > part + part) {
-                secondPart = sdp.slice(part, part + part);
-                thirdPart = sdp.slice(part + part, sdp.length);
-            }
-
-            socket.send({
-                    userToken: self.userToken,
-                    firstPart: firstPart
-                });
-
-            socket.send({
-                    userToken: self.userToken,
-                    secondPart: secondPart
-                });
-
             socket.send({
-                    userToken: self.userToken,
-                    thirdPart: thirdPart
-                });
+                userToken: self.userToken,
+                sdp: JSON.stringify(sdp)
+            });
         }
 
         function socketResponse(response) {
             if (response.userToken == self.userToken) return;
-            if (response.firstPart || response.secondPart || response.thirdPart) {
-                if (response.firstPart) {
-                    inner.firstPart = response.firstPart;
-                    if (inner.secondPart && inner.thirdPart) selfInvoker();
-                }
-                if (response.secondPart) {
-                    inner.secondPart = response.secondPart;
-                    if (inner.firstPart && inner.thirdPart) selfInvoker();
-                }
-
-                if (response.thirdPart) {
-                    inner.thirdPart = response.thirdPart;
-                    if (inner.firstPart && inner.secondPart) selfInvoker();
-                }
+            if (response.sdp) {
+                inner.sdp = JSON.parse(response.sdp);
+                selfInvoker();
             }
 
             if (response.candidate && !gotstream) {
-                peer && peer.addICE({
+                if (!peer) console.error('missed an ice', response.candidate);
+                else
+                    peer.addICE({
                         sdpMLineIndex: response.candidate.sdpMLineIndex,
                         candidate: JSON.parse(response.candidate.candidate)
                     });
             }
+
+            if (response.left) {
+                if (peer && peer.peer) {
+                    peer.peer.close();
+                    peer.peer = null;
+                }
+            }
         }
 
         var invokedOnce = false;
@@ -175,18 +156,39 @@ var conference = function (config) {
 
             invokedOnce = true;
 
-            inner.sdp = JSON.parse(inner.firstPart + inner.secondPart + inner.thirdPart);
             if (isofferer) peer.addAnswerSDP(inner.sdp);
             else initPeer(inner.sdp);
         }
     }
 
+    function leave() {
+        var length = sockets.length;
+        for (var i = 0; i < length; i++) {
+            var socket = sockets[i];
+            if (socket) {
+                socket.send({
+                    left: true,
+                    userToken: self.userToken
+                });
+                delete sockets[i];
+            }
+        }
+    }
+
+    window.onbeforeunload = function() {
+        leave();
+    };
+
+    window.onkeyup = function(e) {
+        if (e.keyCode == 116) leave();
+    };
+
     function startBroadcasting() {
         defaultSocket && defaultSocket.send({
-                roomToken: self.roomToken,
-                roomName: self.roomName,
-                broadcaster: self.userToken
-            });
+            roomToken: self.roomToken,
+            roomName: self.roomName,
+            broadcaster: self.userToken
+        });
         setTimeout(startBroadcasting, 3000);
     }
 
@@ -196,20 +198,19 @@ var conference = function (config) {
 
         var new_channel = uniqueToken();
         openSubSocket({
-                channel: new_channel,
-                closeSocket: true
-            });
+            channel: new_channel
+        });
 
         defaultSocket.send({
-                participant: true,
-                userToken: self.userToken,
-                joinUser: channel,
-                channel: new_channel
-            });
+            participant: true,
+            userToken: self.userToken,
+            joinUser: channel,
+            channel: new_channel
+        });
     }
 
     function uniqueToken() {
-        var s4 = function () {
+        var s4 = function() {
             return Math.floor(Math.random() * 0x10000).toString(16);
         };
         return s4() + s4() + "-" + s4() + "-" + s4() + "-" + s4() + "-" + s4() + s4() + s4();
@@ -217,7 +218,7 @@ var conference = function (config) {
 
     openDefaultSocket();
     return {
-        createRoom: function (_config) {
+        createRoom: function(_config) {
             self.roomName = _config.roomName || 'Anonymous';
             self.roomToken = uniqueToken();
 
@@ -225,19 +226,19 @@ var conference = function (config) {
             isGetNewRoom = false;
             startBroadcasting();
         },
-        joinRoom: function (_config) {
+        joinRoom: function(_config) {
             self.roomToken = _config.roomToken;
             isGetNewRoom = false;
 
             openSubSocket({
-                    channel: self.userToken
-                });
+                channel: self.userToken
+            });
 
             defaultSocket.send({
-                    participant: true,
-                    userToken: self.userToken,
-                    joinUser: _config.joinUser
-                });
+                participant: true,
+                userToken: self.userToken,
+                joinUser: _config.joinUser
+            });
         }
     };
 };
diff --git a/video-conferencing/index.html b/video-conferencing/index.html
index 2ff38a87..5935f78f 100644
--- a/video-conferencing/index.html
+++ b/video-conferencing/index.html
@@ -172,7 +172,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
         <h1>
             WebRTC video-conferencing experiment / <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing" target="_blank">Source Code on Github</a>
@@ -183,7 +183,7 @@ <h1>
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table class="visible">
@@ -210,10 +210,10 @@ <h2>
 
         <div id="participants"></div>
 
-        <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/video-conferencing/conference.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/video-conferencing/conference-ui.js"></script>
+        <script src="https://www.webrtc-experiment.com/socket.io.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
+        <script src="https://www.webrtc-experiment.com/video-conferencing/conference.js"> </script>
+        <script src="https://www.webrtc-experiment.com/video-conferencing/conference-ui.js"></script>
     </article>
 	
     <article>
@@ -241,13 +241,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>
diff --git a/webrtc-broadcasting/README.md b/webrtc-broadcasting/README.md
index 943f6f5c..6c82caaa 100644
--- a/webrtc-broadcasting/README.md
+++ b/webrtc-broadcasting/README.md
@@ -1,4 +1,4 @@
-#### WebRTC One-Way video sharing/broadcasting / [Demo](https://webrtc-experiment.appspot.com/webrtc-broadcasting/)
+#### WebRTC One-Way video sharing/broadcasting / [Demo](https://www.webrtc-experiment.com/webrtc-broadcasting/)
 
 Participants can view your broadcasted video **anonymously**. They can also listen you without allowing access to their own microphone!
 
@@ -10,6 +10,8 @@ You can:
 2. Share you camera in one-way over many peers
 3. Share/transmit your voice in one-way over many peers
 
+=
+
 #### How WebRTC One-Way Broadcasting Works?
 
 It is a **one-to-many** audio/video/screen sharing experiment. However, only room initiator will be asked to allow access to camera/microphone because his media stream will be shared in one-way over all connected peers.
@@ -37,9 +39,11 @@ For users who are watching your video stream anonymously; **2 incoming RTP** por
 
 Again, because it is one-way streaming; **no outgoing RTP ports** will be opened on room participants' side.
 
+=
+
 #### Browser Support
 
-This [WebRTC One-Way Broadcasting](https://webrtc-experiment.appspot.com/webrtc-broadcasting/) experiment works fine on following web-browsers:
+This [WebRTC One-Way Broadcasting](https://www.webrtc-experiment.com/webrtc-broadcasting/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -47,6 +51,8 @@ This [WebRTC One-Way Broadcasting](https://webrtc-experiment.appspot.com/webrtc-
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
+=
+
 #### License
 
-[WebRTC One-Way Broadcasting](https://webrtc-experiment.appspot.com/webrtc-broadcasting/) experiment is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC One-Way Broadcasting](https://www.webrtc-experiment.com/webrtc-broadcasting/) experiment is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/webrtc-broadcasting/broadcast-ui.js b/webrtc-broadcasting/broadcast-ui.js
index 7ad0e629..ea7d4508 100644
--- a/webrtc-broadcasting/broadcast-ui.js
+++ b/webrtc-broadcasting/broadcast-ui.js
@@ -1,17 +1,34 @@
-var config = {
+// 2013, @muazkh » github.com/muaz-khan
+// MIT License » https://webrtc-experiment.appspot.com/licence/
+// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/webrtc-broadcasting
+
+var config = {
     openSocket: function (config) {
-        var channel = config.channel || location.hash.replace('#', '') || 'video-oneway-broadcasting';
-        var socket = new Firebase('https://webrtc.firebaseIO.com/' + channel);
+        var SIGNALING_SERVER = 'https://www.webrtc-experiment.com:8553/',
+            defaultChannel = location.hash.substr(1) || 'video-oneway-broadcasting';
+
+        var channel = config.channel || defaultChannel;
+        var sender = Math.round(Math.random() * 999999999) + 999999999;
+
+        io.connect(SIGNALING_SERVER).emit('new-channel', {
+            channel: channel,
+            sender: sender
+        });
+
+        var socket = io.connect(SIGNALING_SERVER + channel);
         socket.channel = channel;
-        socket.on("child_added", function (data) {
-            config.onmessage && config.onmessage(data.val());
+        socket.on('connect', function() {
+            if (config.callback) config.callback(socket);
         });
-        socket.send = function(data) {
-            this.push(data);
+
+        socket.send = function(message) {
+            socket.emit('message', {
+                sender: sender,
+                data: message
+            });
         };
-        config.onopen && setTimeout(config.onopen, 1);
-        socket.onDisconnect().remove();
-        return socket;
+
+        socket.on('message', config.onmessage);
     },
     onRemoteStream: function (htmlElement) {
         htmlElement.setAttribute('controls', true);
@@ -33,7 +50,7 @@
             if (room.isAudio) tr.setAttribute('accesskey', room.isAudio);
 
             tr.innerHTML = '<td>' + room.roomName + '</td>' +
-                '<td><button class="join" id="' + room.roomToken + '">Join</button></td>';
+                '<td><button class="join" id="' + room.roomToken + '">View Anonymously</button></td>';
             roomsList.insertBefore(tr, roomsList.firstChild);
 
             tr.onclick = function () {
diff --git a/webrtc-broadcasting/index.html b/webrtc-broadcasting/index.html
index 1e5e1b09..091af9c3 100644
--- a/webrtc-broadcasting/index.html
+++ b/webrtc-broadcasting/index.html
@@ -129,7 +129,7 @@
     
     <body>
         <article>
-            <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+            <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
             <h1>
 				WebRTC one-way audio, video and screen sharing
@@ -140,7 +140,7 @@ <h1>
 			</p>
 			
 			<section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
 			
             <table class="visible">
@@ -171,10 +171,10 @@ <h2>
                 </tr>
             </table>
             <div id="participants"></div>
-            <script src="https://webrtc-experiment.appspot.com/firebase.js"></script>
-			<script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"></script>
-            <script src="https://webrtc-experiment.appspot.com/webrtc-broadcasting/broadcast.js"></script>
-            <script src="https://webrtc-experiment.appspot.com/webrtc-broadcasting/broadcast-ui.js"></script>
+            <script src="https://www.webrtc-experiment.com/socket.io.js"></script>
+			<script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
+            <script src="https://www.webrtc-experiment.com/webrtc-broadcasting/broadcast.js"></script>
+            <script src="https://www.webrtc-experiment.com/webrtc-broadcasting/broadcast-ui.js"></script>
             
             <ol>
                 <li>Share your screen in one-way over many peers</li>
@@ -205,14 +205,14 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
         </article>
         <footer>
             <p>
-                <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+                <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span> 2013 </span> »
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
             </p>
         </footer>
-        <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+        <script src="https://www.webrtc-experiment.com/common.js"></script>
     </body>
 
 </html>
\ No newline at end of file
diff --git a/websocket-over-nodejs/README.md b/websocket-over-nodejs/README.md
index f6671fd7..98e49cfc 100644
--- a/websocket-over-nodejs/README.md
+++ b/websocket-over-nodejs/README.md
@@ -12,7 +12,7 @@ This experiment is using **WebSocket over Node.js** for signaling. Follow these
 
 `http://localhost:9999/` will be auto-opened.
 
-----
+=
 
 #### If you want to deploy your application
 
@@ -22,7 +22,7 @@ This experiment is using **WebSocket over Node.js** for signaling. Follow these
 
 and you're done!
 
-----
+=
 
 #### How to use?
 
@@ -55,7 +55,7 @@ openSignalingChannel: function(config) {
 }
 ```
 
-----
+=
 
 #### Presence Detection
 
@@ -83,7 +83,7 @@ function testChannelPresence(channel) {
 testChannelPresence('default-channel');
 ```
 
-----
+=
 
 #### Ports
 
@@ -110,7 +110,7 @@ var SIGNALING_SERVER = 'ws://' + document.domain + ':8888/';
 var websocket = new WebSocket(SIGNALING_SERVER);
 ```
 
-----
+=
 
 #### if fails to run....
 
@@ -131,19 +131,19 @@ childProcess.spawn(chromeURL, ['-incognito', openURL]);
 
 Actually, these lines are auto-opening chrome/firefox instances.
 
-----
+=
 
 #### What is `cd folder-location`?
 
 Using this command; you can open project's directory (i.e. folder) in a command prompt window.
 
-----
+=
 
 #### What is `node signaler`?
 
 This command runs the `signaler.js` file as a Node.js instance. That file handles `WebSocket over Node.js` server side stuff.
 
-----
+=
 
 #### What is `jitsu deploy`?
 
@@ -155,7 +155,7 @@ http://username.jit.su/
 
 See the demo URL: http://websocket-over-nodejs.jit.su/
 
-----
+=
 
 #### Are you beginner or totally novice?
 
@@ -164,13 +164,13 @@ See the demo URL: http://websocket-over-nodejs.jit.su/
 3. You can use same command prompt to run any `node.js` file; also you can write `nodejitsu` commands in the same place e.g. `jitsu deploy` or `jitsu login` etc.
 4. Default port `9999` is used for this experiment. You can manually open this URL: `http://localhost:9999/`
 
-----
+=
 
 #### Signaling Concepts
 
 Interested to understand WebRTC Signaling Concepts? Read [this document](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/socketio-over-nodejs/Signaling-Concepts.md).
 
-----
+=
 
 #### License
 
diff --git a/websocket-over-nodejs/video-conferencing/RTCPeerConnection-v1.5.js b/websocket-over-nodejs/video-conferencing/RTCPeerConnection-v1.5.js
index 980fd0f4..41f0ba82 100644
--- a/websocket-over-nodejs/video-conferencing/RTCPeerConnection-v1.5.js
+++ b/websocket-over-nodejs/video-conferencing/RTCPeerConnection-v1.5.js
@@ -1,35 +1,37 @@
-/*  MIT License: https://webrtc-experiment.appspot.com/licence/ 
- 2013, Muaz Khan<muazkh>--[github.com/muaz-khan]
+// 2013, @muazkh - github.com/muaz-khan
+// MIT License - https://webrtc-experiment.appspot.com/licence/
+// Documentation - https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCPeerConnection
 
- Demo & Documentation: http://bit.ly/RTCPeerConnection-Documentation */
-window.moz = !! navigator.mozGetUserMedia;
-var RTCPeerConnection = function (options) {
+window.moz = !!navigator.mozGetUserMedia;
+
+function RTCPeerConnection(options) {
     var w = window,
         PeerConnection = w.mozRTCPeerConnection || w.webkitRTCPeerConnection,
         SessionDescription = w.mozRTCSessionDescription || w.RTCSessionDescription,
         IceCandidate = w.mozRTCIceCandidate || w.RTCIceCandidate;
 
-    STUN = {
+    var STUN = {
         url: !moz ? 'stun:stun.l.google.com:19302' : 'stun:23.21.150.121'
     };
 
-    TURN = {
-        url: 'turn:webrtc%40live.com@numb.viagenie.ca',
-        credential: 'muazkh'
+    var TURN = {
+        url: 'turn:homeo@turn.bistri.com:80',
+        credential: 'homeo'
     };
 
-    iceServers = {
+    var iceServers = {
         iceServers: options.iceServers || [STUN]
     };
 
     if (!moz && !options.iceServers) {
-        if (parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2]) >= 28)
+        if (parseInt(navigator.userAgent.match( /Chrom(e|ium)\/([0-9]+)\./ )[2]) >= 28)
             TURN = {
-                url: 'turn:numb.viagenie.ca',
-                credential: 'muazkh',
-                username: 'webrtc@live.com'
-        };
-        iceServers.iceServers = [TURN, STUN];
+                url: 'turn:turn.bistri.com:80',
+                credential: 'homeo',
+                username: 'homeo'
+            };
+
+        iceServers.iceServers = [STUN, TURN];
     }
 
     var optional = {
@@ -38,38 +40,50 @@ var RTCPeerConnection = function (options) {
 
     if (!moz) {
         optional.optional = [{
-                DtlsSrtpKeyAgreement: true
-            }
-        ];
+            DtlsSrtpKeyAgreement: true
+        }];
 
         if (options.onChannelMessage)
             optional.optional = [{
-                    RtpDataChannels: true
-                }
-            ];
+                RtpDataChannels: true
+            }];
     }
 
-    var peerConnection = new PeerConnection(iceServers, optional);
+    var peer = new PeerConnection(iceServers, optional);
 
     openOffererChannel();
 
-    peerConnection.onicecandidate = onicecandidate;
-    if (options.attachStream) peerConnection.addStream(options.attachStream);
-    peerConnection.onaddstream = onaddstream;
+    peer.onicecandidate = function(event) {
+        if (event.candidate)
+            options.onICE(event.candidate);
+    };
 
-    function onicecandidate(event) {
-        if (!event.candidate || !peerConnection) return;
-        if (options.onICE) options.onICE(event.candidate);
+    // attachStream = MediaStream;
+    if (options.attachStream) peer.addStream(options.attachStream);
+
+    // attachStreams[0] = audio-stream;
+    // attachStreams[1] = video-stream;
+    // attachStreams[2] = screen-capturing-stream;
+    if (options.attachStreams && options.attachStream.length) {
+        var streams = options.attachStreams;
+        for (var i = 0; i < streams.length; i++) {
+            peer.addStream(streams[i]);
+        }
     }
 
-    var remoteStreamEventFired = false;
+    peer.onaddstream = function(event) {
+        var remoteMediaStream = event.stream;
 
-    function onaddstream(event) {
-        info('------------onaddstream');
-        if (remoteStreamEventFired || !event || !options.onRemoteStream) return;
-        remoteStreamEventFired = true;
-        options.onRemoteStream(event.stream);
-    }
+        // onRemoteStreamEnded(MediaStream)
+        remoteMediaStream.onended = function() {
+            if (options.onRemoteStreamEnded) options.onRemoteStreamEnded(remoteMediaStream);
+        };
+
+        // onRemoteStream(MediaStream)
+        if (options.onRemoteStream) options.onRemoteStream(remoteMediaStream);
+
+        console.debug('on:add:stream', remoteMediaStream);
+    };
 
     var constraints = options.constraints || {
         optional: [],
@@ -79,168 +93,175 @@ var RTCPeerConnection = function (options) {
         }
     };
 
-    var chars = 'ABCDEFGHIJKLMNOPQRSTUVWXYZ'.split(''),
-        extractedChars = '';
+    // onOfferSDP(RTCSessionDescription)
 
-    function getChars() {
-        extractedChars += chars[parseInt(Math.random() * 40)] || '';
-        if (extractedChars.length < 40) getChars();
+    function createOffer() {
+        if (!options.onOfferSDP) return;
 
-        return extractedChars;
+        peer.createOffer(function(sessionDescription) {
+            sessionDescription.sdp = serializeSdp(sessionDescription.sdp);
+            peer.setLocalDescription(sessionDescription);
+            options.onOfferSDP(sessionDescription);
+        }, null, constraints);
     }
 
-    function getInteropSDP(sdp) {
-        // for audio-only streaming: multiple-crypto lines are not allowed
-        if (options.onAnswerSDP)
-            sdp = sdp.replace(/(a=crypto:0 AES_CM_128_HMAC_SHA1_32)(.*?)(\r\n)/g, '');
+    // onAnswerSDP(RTCSessionDescription)
 
+    function createAnswer() {
+        if (!options.onAnswerSDP) return;
 
-        var inline = getChars() + '\r\n' + (extractedChars = '');
-        sdp = sdp.indexOf('a=crypto') == -1 ? sdp.replace(/c=IN/g,
-            'a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:' + inline +
-            'c=IN') : sdp;
+        peer.setRemoteDescription(new SessionDescription(options.offerSDP));
+        peer.createAnswer(function(sessionDescription) {
+            sessionDescription.sdp = serializeSdp(sessionDescription.sdp);
+            peer.setLocalDescription(sessionDescription);
+            options.onAnswerSDP(sessionDescription);
+        }, null, constraints);
+    }
 
-        if (options.offerSDP) {
-            info('\n--------offer sdp provided by offerer\n');
-            info(options.offerSDP.sdp);
-        }
+    // if Mozilla Firefox & DataChannel; offer/answer will be created later
+    if ((options.onChannelMessage && !moz) || !options.onChannelMessage) {
+        createOffer();
+        createAnswer();
+    }
 
-        info(options.onOfferSDP ? '\n--------offer\n' : '\n--------answer\n');
-        info('sdp: ' + sdp);
+
+    // DataChannel Bandwidth
+
+    function setBandwidth(sdp) {
+        // remove existing bandwidth lines
+        sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , '');
+        sdp = sdp.replace( /a=mid:data\r\n/g , 'a=mid:data\r\nb=AS:1638400\r\n');
 
         return sdp;
     }
 
-    if (moz && !options.onChannelMessage) constraints.mandatory.MozDontOfferDataChannel = true;
+    // old: FF<>Chrome interoperability management
 
-    function createOffer() {
-        if (!options.onOfferSDP) return;
+    function getInteropSDP(sdp) {
+        var chars = 'ABCDEFGHIJKLMNOPQRSTUVWXYZ'.split(''),
+            extractedChars = '';
 
-        peerConnection.createOffer(function (sessionDescription) {
-                sessionDescription.sdp = getInteropSDP(sessionDescription.sdp);
-                peerConnection.setLocalDescription(sessionDescription);
-                options.onOfferSDP(sessionDescription);
-            }, null, constraints);
-    }
+        function getChars() {
+            extractedChars += chars[parseInt(Math.random() * 40)] || '';
+            if (extractedChars.length < 40)
+                getChars();
 
-    function createAnswer() {
-        if (!options.onAnswerSDP) return;
+            return extractedChars;
+        }
 
-        options.offerSDP = new SessionDescription(options.offerSDP);
-        peerConnection.setRemoteDescription(options.offerSDP);
+        // for audio-only streaming: multiple-crypto lines are not allowed
+        if (options.onAnswerSDP)
+            sdp = sdp.replace( /(a=crypto:0 AES_CM_128_HMAC_SHA1_32)(.*?)(\r\n)/g , '');
 
-        peerConnection.createAnswer(function (sessionDescription) {
-                sessionDescription.sdp = getInteropSDP(sessionDescription.sdp);
-                peerConnection.setLocalDescription(sessionDescription);
-                options.onAnswerSDP(sessionDescription);
-            }, null, constraints);
+        var inline = getChars() + '\r\n' + (extractedChars = '');
+        sdp = sdp.indexOf('a=crypto') == -1 ? sdp.replace( /c=IN/g ,
+            'a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:' + inline +
+                'c=IN') : sdp;
+
+        return sdp;
     }
 
-    if ((options.onChannelMessage && !moz) || !options.onChannelMessage) {
-        createOffer();
-        createAnswer();
+    function serializeSdp(sdp) {
+        if (!moz) sdp = setBandwidth(sdp);
+        sdp = getInteropSDP(sdp);
+        console.debug(sdp);
+        return sdp;
     }
 
+    // DataChannel management
     var channel;
 
     function openOffererChannel() {
-        if (!options.onChannelMessage || (moz && !options.onOfferSDP)) return;
+        if (!options.onChannelMessage || (moz && !options.onOfferSDP))
+            return;
 
         _openOffererChannel();
 
-        if (moz && !options.attachStream) {
+        if (moz) {
             navigator.mozGetUserMedia({
                     audio: true,
                     fake: true
-                }, function (stream) {
-                    peerConnection.addStream(stream);
+                }, function(stream) {
+                    peer.addStream(stream);
                     createOffer();
                 }, useless);
         }
     }
 
     function _openOffererChannel() {
-        channel = peerConnection.createDataChannel(
-            options.channel || 'RTCDataChannel',
-            moz ? {} : {
-                reliable: false
-            });
+        channel = peer.createDataChannel(options.channel || 'RTCDataChannel', moz ? { } : {
+            reliable: false
+        });
 
-        if (moz) channel.binaryType = 'blob';
+        if (moz)
+            channel.binaryType = 'blob';
         setChannelEvents();
     }
 
     function setChannelEvents() {
-        channel.onmessage = function (event) {
+        channel.onmessage = function(event) {
             if (options.onChannelMessage) options.onChannelMessage(event);
         };
 
-        channel.onopen = function () {
+        channel.onopen = function() {
             if (options.onChannelOpened) options.onChannelOpened(channel);
         };
-        channel.onclose = function (event) {
+        channel.onclose = function(event) {
             if (options.onChannelClosed) options.onChannelClosed(event);
 
             console.warn('WebRTC DataChannel closed', event);
         };
-        channel.onerror = function (event) {
-            console.error(event);
+        channel.onerror = function(event) {
             if (options.onChannelError) options.onChannelError(event);
         };
     }
 
-    if (options.onAnswerSDP && moz) openAnswererChannel();
+    if (options.onAnswerSDP && moz && options.onChannelMessage)
+        openAnswererChannel();
 
     function openAnswererChannel() {
-        peerConnection.ondatachannel = function (_channel) {
-            channel = _channel;
+        peer.ondatachannel = function(event) {
+            channel = event.channel;
             channel.binaryType = 'blob';
             setChannelEvents();
         };
 
-        if (moz && !options.attachStream) {
+        if (moz) {
             navigator.mozGetUserMedia({
                     audio: true,
                     fake: true
-                }, function (stream) {
-                    peerConnection.addStream(stream);
+                }, function(stream) {
+                    peer.addStream(stream);
                     createAnswer();
                 }, useless);
         }
     }
 
-    function useless() {}
-
-    function info(information) {
-        console.log(information);
+    function useless() {
     }
 
     return {
-        addAnswerSDP: function (sdp) {
-            info('--------adding answer sdp:');
-            info(sdp.sdp);
-
-            sdp = new SessionDescription(sdp);
-            peerConnection.setRemoteDescription(sdp);
+        addAnswerSDP: function(sdp) {
+            peer.setRemoteDescription(new SessionDescription(sdp));
         },
-        addICE: function (candidate) {
-            info(candidate.candidate);
-            peerConnection.addIceCandidate(new IceCandidate({
-                        sdpMLineIndex: candidate.sdpMLineIndex,
-                        candidate: candidate.candidate
-                    }));
+        addICE: function(candidate) {
+            peer.addIceCandidate(new IceCandidate({
+                sdpMLineIndex: candidate.sdpMLineIndex,
+                candidate: candidate.candidate
+            }));
         },
 
-        peer: peerConnection,
+        peer: peer,
         channel: channel,
-        sendData: function (message) {
+        sendData: function(message) {
             channel && channel.send(message);
         }
     };
-};
+}
 
+// getUserMedia
 var video_constraints = {
-    mandatory: {},
+    mandatory: { },
     optional: []
 };
 
@@ -251,7 +272,7 @@ function getUserMedia(options) {
     n.getMedia(options.constraints || {
             audio: true,
             video: video_constraints
-        }, streaming, options.onerror || function (e) {
+        }, streaming, options.onerror || function(e) {
             console.error(e);
         });
 
diff --git a/websocket-over-nodejs/video-conferencing/index.html b/websocket-over-nodejs/video-conferencing/index.html
index 94a9eaa5..90e418fd 100644
--- a/websocket-over-nodejs/video-conferencing/index.html
+++ b/websocket-over-nodejs/video-conferencing/index.html
@@ -172,14 +172,14 @@
 <body>
     <article> 
     
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
       <h1>
         WebRTC video-conferencing & <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/websocket-over-nodejs" target="_blank">WebSocket over Node.js</a>
       </h1>
 
             <p> <span>Copyright © 2013</span> <a href="https://github.com/muaz-khan" target="_blank">Muaz Khan</a><span>&lt;</span><a href="http://twitter.com/muazkh" target="_blank">@muazkh</a><span>&gt;.</span></p>
             <section class="plusone-gplus">
-                <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+                <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
             </section>
             <table class="visible">
                 <tr>
@@ -224,11 +224,11 @@ <h2 style="font-size:1em;">
             </section>
         </article>
         <footer>
-            <p> <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
+            <p> <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>© <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>» <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a
                 href="http://twitter.com/muazkh" target="_blank">@muazkh</a>» <a href="https://github.com/muaz-khan" target="_blank">Github</a>
             </p>
         </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"></script>
+    <script src="https://www.webrtc-experiment.com/common.js"></script>
 </body>
 </html>
\ No newline at end of file
diff --git a/websocket/README.md b/websocket/README.md
index 767d267e..810a4441 100644
--- a/websocket/README.md
+++ b/websocket/README.md
@@ -1,8 +1,8 @@
-#### WebRTC One-to-One video sharing using WebSockets / [Demo](https://webrtc-experiment.appspot.com/websocket/)
+#### WebRTC One-to-One video sharing using WebSockets / [Demo](https://www.webrtc-experiment.com/websocket/)
 
 This `WebRTC Experiment` is using `WebSockets` for signalig.
 
-----
+=
 
 #### How to use your own WebSocket implementation?
 
@@ -37,11 +37,11 @@ openSignalingChannel: function(config) {
 
 In `ui.js` files you can find `openSocket` method; or in all libraries; you can find `openSignalingChannel` method.
 
-----
+=
 
 #### Browser Support
 
-This [One-to-one WebRTC video chat using WebSocket](https://webrtc-experiment.appspot.com/websocket/) experiment works fine on following web-browsers:
+This [One-to-one WebRTC video chat using WebSocket](https://www.webrtc-experiment.com/websocket/) experiment works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -49,8 +49,8 @@ This [One-to-one WebRTC video chat using WebSocket](https://webrtc-experiment.ap
 | Google Chrome | [Stable](https://www.google.com/intl/en_uk/chrome/browser/) / [Canary](https://www.google.com/intl/en/chrome/browser/canary.html) / [Beta](https://www.google.com/intl/en/chrome/browser/beta.html) / [Dev](https://www.google.com/intl/en/chrome/browser/index.html?extra=devchannel#eula) |
 | Android | [Chrome Beta](https://play.google.com/store/apps/details?id=com.chrome.beta&hl=en) |
 
-----
+=
 
 #### License
 
-[WebRTC One-to-One video sharing using WebSockets](https://webrtc-experiment.appspot.com/websocket/) is released under [MIT licence](https://webrtc-experiment.appspot.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
+[WebRTC One-to-One video sharing using WebSockets](https://www.webrtc-experiment.com/websocket/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) 2013 [Muaz Khan](https://plus.google.com/100325991024054712503).
diff --git a/websocket/index.html b/websocket/index.html
index 8beca1e6..b083ad04 100644
--- a/websocket/index.html
+++ b/websocket/index.html
@@ -172,7 +172,7 @@
 
 <body>
     <article>
-        <a href="https://webrtc-experiment.appspot.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
+        <a href="https://www.webrtc-experiment.com/" style="border-bottom: 1px solid #2844FA; font-size: 1.2em; position: absolute; text-decoration: none;right:0;top:0;">↑ WEBRTC EXPERIMENTS</a>
 
         <h1>Using WebSockets for signaling / <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/websocket" target="_blank">Source Code on Github</a></h1>
         <p>
@@ -181,7 +181,7 @@ <h1>Using WebSockets for signaling / <a href="https://github.com/muaz-khan/WebRT
         </p>
 
         <section class="plusone-gplus">
-            <div class="g-plusone" data-href="https://webrtc-experiment.appspot.com/"></div>
+            <div class="g-plusone" data-href="https://www.webrtc-experiment.com/"></div>
         </section>
 
         <table class="visible">
@@ -208,10 +208,10 @@ <h2>
 
         <div id="participants"></div>
 
-        <script src="https://webrtc-experiment.appspot.com/dependencies/websocket.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/RTCPeerConnection-v1.5.js"></script>
-        <script src="https://webrtc-experiment.appspot.com/websocket/rtclib.js"> </script>
-        <script src="https://webrtc-experiment.appspot.com/websocket/ui.js"></script>
+        <script src="https://www.webrtc-experiment.com/dependencies/websocket.js"></script>
+        <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"></script>
+        <script src="https://www.webrtc-experiment.com/websocket/rtclib.js"> </script>
+        <script src="https://www.webrtc-experiment.com/websocket/ui.js"></script>
 		<p>WebRTC one-to-one video sharing experiment where PubNub's WebSocket implementation is used for signaling.</p>
 	</article>
 	<article>
@@ -234,13 +234,13 @@ <h2 id="feedback" style="padding: .2em .4em; border-bottom: 1px solid rgb(189, 1
     </article>
     <footer>
         <p>
-            <a href="https://webrtc-experiment.appspot.com/" target="_blank">WebRTC Experiments!</a>©
+            <a href="https://www.webrtc-experiment.com/" target="_blank">WebRTC Experiments!</a>©
                 <a href="https://plus.google.com/100325991024054712503" rel="author" target="_blank">Muaz Khan</a>, <span>2013 </span>»
                 <a href="mailto:muazkh@gmail.com" target="_blank">Email</a>»
                 <a href="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
         </p>
     </footer>
-    <script src="https://webrtc-experiment.appspot.com/common.js"> </script>
+    <script src="https://www.webrtc-experiment.com/common.js"> </script>
 </body>
 </html>