diff --git a/Chrome-Extensions/desktopCapture/README.md b/Chrome-Extensions/desktopCapture/README.md
index ceaeb0e8..5fc66eb8 100644
--- a/Chrome-Extensions/desktopCapture/README.md
+++ b/Chrome-Extensions/desktopCapture/README.md
@@ -253,7 +253,7 @@ Ref: http://kurtextrem.github.io/ChromiumFlags/#allow-http-screen-capture
 
 ## Browser Support
 
-[Capture Screen Extension](https://www.webrtc-experiment.com/store/capture-screen/) works fine on following web-browsers:
+[Capture Screen Extension](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Chrome-Extensions/desktopCapture) works fine on following web-browsers:
 
 | Browser        | Support           |
 | ------------- |-------------|
@@ -263,4 +263,4 @@ Ref: http://kurtextrem.github.io/ChromiumFlags/#allow-http-screen-capture
 
 #### License
 
-[Capture Screen Extension](https://www.webrtc-experiment.com/store/capture-screen/) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) [Muaz Khan](https://plus.google.com/+MuazKhan).
+[Capture Screen Extension](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Chrome-Extensions/desktopCapture) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) [Muaz Khan](https://plus.google.com/+MuazKhan).
diff --git a/DataChannel/README.md b/DataChannel/README.md
index 6e2a8147..a2dc247c 100644
--- a/DataChannel/README.md
+++ b/DataChannel/README.md
@@ -23,7 +23,7 @@ DataChannel.js is a JavaScript library useful to write many-to-many i.e. group f
 ##### Try a Quick Demo (Text Chat)
 
 ```html
-<script src="https://www.webrtc-experiment.com/DataChannel.js"> </script>
+<script src="//cdn.webrtc-experiment.com/DataChannel.js"> </script>
 <button id="setup-datachannel" style="width:30%;">Open NEW DataChannel</button>
 <input type="text" id="chat-input" disabled style="font-size: 2em; width: 65%;"><br />
 <div id="chat-output"></div>
@@ -69,7 +69,7 @@ document.querySelector('button#setup-datachannel').onclick = function () {
 ##### First Step: Link the library
 
 ```html
-<script src="https://www.webrtc-experiment.com/DataChannel.js"></script>
+<script src="//cdn.webrtc-experiment.com/DataChannel.js"></script>
 ```
 
 =
diff --git a/DataChannel/auto-session-establishment.html b/DataChannel/auto-session-establishment.html
index 6e15d76b..ae765563 100644
--- a/DataChannel/auto-session-establishment.html
+++ b/DataChannel/auto-session-establishment.html
@@ -9,6 +9,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         <script>
             var hash = window.location.hash.replace('#', '');
             if (!hash.length) {
@@ -17,149 +18,6 @@
             }
         </script>
         <style>
-            @import url(https://fonts.googleapis.com/css?family=Inconsolata);
-
-            html { background: #eee; }
-
-            body {
-                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;
-                max-width: 900px;
-                min-width: 360px;
-                width: 80%;
-            }
-
-            article {
-                background: #fff;
-                border: 1px solid;
-                border-color: #ddd #aaa #aaa #ddd;
-                margin: 2.5em auto 0 auto;
-                padding: 2em;
-            }
-
-            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;
-                font-size: inherit;
-                padding: 1em;
-                text-decoration: none;
-                text-shadow: 0 1px 1px #fff;
-            }
-
-            footer a:hover, footer a:focus { color: #111; }
-
-            h1, h2 {
-                border-bottom: 1px solid black;
-                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; }
-
-            :-moz-any-link:focus {
-                border: 0;
-                color: #000;
-            }
-
-            ::selection { background: #ccc; }
-
-            ::-moz-selection { background: #ccc; }
-
-            button, select {
-                -moz-border-radius: 3px;
-                -moz-transition: none;
-                -webkit-transition: none;
-                background: #0370ea;
-                background: -moz-linear-gradient(top, #008dfd 0, #0370ea 100%);
-                background: -webkit-linear-gradient(top, #008dfd 0, #0370ea 100%);
-                border: 1px solid #076bd2;
-                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;
-            }
-
-            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;
-            }
-
-            select {
-                color: black;
-                text-shadow: none;
-            }
-
-            td { vertical-align: top; }
-
-            #chat-output div, #file-progress div {
-                border: 1px solid black;
-                border-bottom: 0;
-                padding: .1em .4em;
-            }
-
-            input {
-                border: 1px solid black;
-                font-family: inherit;
-                margin: .1em .3em;
-                outline: none;
-                padding: .1em .2em;
-                width: 97%;
-            }
-
-            #chat-output, #file-progress {
-                margin: 0 0 0 .4em;
-                max-height: 12em;
-                overflow: auto;
-            }
-
-            pre {
-                border-left: 2px solid red;
-                margin-left: 2em;
-                padding-left: 1em;
-            }
-
             section.user-id {
                 float: left;
                 overflow: hidden;
@@ -182,8 +40,8 @@
             document.createElement('footer');
         </script>
 
-        <script src="https://www.webrtc-experiment.com/DataChannel.js"> </script>
-        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/DataChannel.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
     </head>
 
     <body>
@@ -195,16 +53,29 @@
 
             <h1>Auto Session Establishment using <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel"
                                                     target="_blank">DataChannel.js</a>
+            
             </h1>
-
-            <p>
-                <span>Copyright © 2014</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://www.webrtc-experiment.com/"></div>
-            </section>
+            
+            <div class="github-stargazers"></div>
+
+                <p>
+                    <a href="https://www.webrtc-experiment.com/">HOME</a>
+                    <span> &copy; </span>
+                    <a href="http://www.MuazKhan.com/" target="_blank">Muaz Khan</a>
+                    
+                    .
+                    <a href="http://twitter.com/WebRTCWeb" target="_blank" title="Twitter profile for WebRTC Experiments">@WebRTCWeb</a>
+                    
+                    .
+                    <a href="https://github.com/muaz-khan?tab=repositories" target="_blank" title="Github Profile">Github</a>
+                    
+                    .
+                    <a href="https://github.com/muaz-khan/WebRTC-Experiment/issues?state=open" target="_blank">Latest issues</a>
+                    
+                    .
+                    <a href="https://github.com/muaz-khan/WebRTC-Experiment/commits/master" target="_blank">What's New?</a>
+                </p>
+            
             <table style="border-left: 1px solid black; width: 100%;">
                 <tr>
                     <td>
@@ -214,7 +85,7 @@ <h2 style="display: block; font-size: 1em; text-align: center;">Text Chat</h2>
 
                         <input type="text" id="user-id" style="font-size: 1.2em; margin-right: 0; width: 5em;" placeholder="all"
                                disabled title="Enter user-id to send direct messages.">
-                        <input type="text" id="chat-input" style="font-size: 1.2em; margin-left: -.5em; width: 20em;"
+                        <input type="text" id="chat-input" style="font-size: 1.2em; margin-left: -.5em; width: 18em;"
                                placeholder="chat message" disabled>
                     </td>
                     <td style="background: white; border-left: 1px solid black;">
@@ -316,7 +187,7 @@ <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://www.webrtc-experiment.com/DataChannel.js" target="_blank">//www.webrtc-experiment.com/DataChannel.js</a>"&gt;&lt;/script&gt;
+&lt;script src="//cdn.webrtc-experiment.com/DataChannel.js"&gt;&lt;/script&gt;
 &lt;script&gt;
     var channel = new <strong>DataChannel</strong>('default-channel');
 
@@ -462,17 +333,22 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
                 </div>
                 <button id="send-message" style="font-size: 1em;">Send Message</button>
             </section>
+            
+            <section class="experiment own-widgets latest-commits">
+                <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a href="https://github.com/muaz-khan/WebRTC-Experiment/commits/master" target="_blank">Latest Updates</a></h2>
+                <div id="github-commits"></div>
+            </section>
         </article>
+        
+        <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel" class="fork-left"></a>
+        
         <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>2014 </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="https://www.webrtc-experiment.com/">WebRTC Experiments</a>
+                © <a href="https://plus.google.com/+MuazKhan" rel="author" target="_blank">Muaz Khan</a>
+                <a href="mailto:muazkh@gmail.com" target="_blank">muazkh@gmail.com</a>
             </p>
         </footer>
-        <script src="https://www.webrtc-experiment.com/common.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js"> </script>
     </body>
 </html>
diff --git a/DataChannel/index.html b/DataChannel/index.html
index 845a253b..599bebe2 100644
--- a/DataChannel/index.html
+++ b/DataChannel/index.html
@@ -14,7 +14,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         <style>
             #chat-output div, #file-progress div {
                 border: 1px solid black;
@@ -44,8 +44,8 @@
         </script>
 		
         <!-- DataChannel.js library -->
-        <script src="https://www.webrtc-experiment.com/DataChannel.js"> </script>
-        <script src="https://www.webrtc-experiment.com/socket.io.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/DataChannel.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/socket.io.js"> </script>
     </head>
 
     <body>
@@ -276,7 +276,7 @@ <h2 class="header">DataChannel.js Features:</h2>
             <section class="experiment">
                 <h2 class="header">How to use DataChannel.js?</h2>
                 <pre>
-&lt;script src="https://www.webrtc-experiment.com/DataChannel.js"&gt; &lt;/script&gt;
+&lt;script src="//cdn.webrtc-experiment.com/DataChannel.js"&gt; &lt;/script&gt;
 
 &lt;input type="text" id="chat-input" disabled 
        style="font-size: 2em; width: 98%;"&gt;&lt;br /&gt;
@@ -316,7 +316,8 @@ <h2 class="header">How to use DataChannel.js?</h2>
 </pre>
             </section>
         </article>
-        <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel" class="fork-right"></a>
+        
+        <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel" class="fork-left"></a>
         
         <footer>
             <p>
@@ -327,6 +328,6 @@ <h2 class="header">How to use DataChannel.js?</h2>
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/DataChannel/simple.html b/DataChannel/simple.html
index 1323cc72..15738e96 100644
--- a/DataChannel/simple.html
+++ b/DataChannel/simple.html
@@ -9,121 +9,10 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
+        
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
+        
         <style>
-            @import url(https://fonts.googleapis.com/css?family=Inconsolata);
-
-            html { background: #eee; }
-
-            body {
-                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;
-                max-width: 900px;
-                min-width: 360px;
-                width: 80%;
-            }
-
-            article {
-                background: #fff;
-                border: 1px solid;
-                border-color: #ddd #aaa #aaa #ddd;
-                margin: 2.5em auto 0 auto;
-                padding: 2em;
-            }
-
-            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;
-                font-size: inherit;
-                padding: 1em;
-                text-decoration: none;
-                text-shadow: 0 1px 1px #fff;
-            }
-
-            footer a:hover, footer a:focus { color: #111; }
-
-            h1, h2 {
-                border-bottom: 1px solid black;
-                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; }
-
-            :-moz-any-link:focus {
-                border: 0;
-                color: #000;
-            }
-
-            ::selection { background: #ccc; }
-
-            ::-moz-selection { background: #ccc; }
-
-            button, select {
-                -moz-border-radius: 3px;
-                -moz-transition: none;
-                -webkit-transition: none;
-                background: #0370ea;
-                background: -moz-linear-gradient(top, #008dfd 0, #0370ea 100%);
-                background: -webkit-linear-gradient(top, #008dfd 0, #0370ea 100%);
-                border: 1px solid #076bd2;
-                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;
-            }
-
-            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;
-            }
-
-            select {
-                color: black;
-                text-shadow: none;
-            }
-
             td { vertical-align: top; }
 
             #chat-output div, #file-progress div {
@@ -159,8 +48,8 @@
             document.createElement('footer');
         </script>
 
-        <script src="https://www.webrtc-experiment.com/DataChannel.js"> </script>
-        <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/DataChannel.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
     </head>
 
     <body>
@@ -174,15 +63,27 @@ <h1>File Sharing + Text Chat using WebRTC <a
                                                           href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel"
                                                           target="_blank">DataChannel</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://www.webrtc-experiment.com/"></div>
-            </section>
+            
+            <div class="github-stargazers"></div>
+
+                <p>
+                    <a href="https://www.webrtc-experiment.com/">HOME</a>
+                    <span> &copy; </span>
+                    <a href="http://www.MuazKhan.com/" target="_blank">Muaz Khan</a>
+                    
+                    .
+                    <a href="http://twitter.com/WebRTCWeb" target="_blank" title="Twitter profile for WebRTC Experiments">@WebRTCWeb</a>
+                    
+                    .
+                    <a href="https://github.com/muaz-khan?tab=repositories" target="_blank" title="Github Profile">Github</a>
+                    
+                    .
+                    <a href="https://github.com/muaz-khan/WebRTC-Experiment/issues?state=open" target="_blank">Latest issues</a>
+                    
+                    .
+                    <a href="https://github.com/muaz-khan/WebRTC-Experiment/commits/master" target="_blank">What's New?</a>
+                </p>
+            
             <section>
                 <h2>Open Data Channel</h2>
                 <input type="text" id="channel" value="channel" style="font-size: 1.1em; text-align: right; width: 4em;"
@@ -301,7 +202,7 @@ <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://www.webrtc-experiment.com/DataChannel.js" target="_blank">https://www.webrtc-experiment.com/DataChannel.js</a>"&gt;&lt;/script&gt;
+&lt;script src="//cdn.webrtc-experiment.com/DataChannel.js"&gt;&lt;/script&gt;
 &lt;script&gt;
     var channel = new <strong>DataChannel</strong>();
 
@@ -412,17 +313,22 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
                 </div>
                 <button id="send-message" style="font-size: 1em;">Send Message</button>
             </section>
+            
+            <section class="experiment own-widgets latest-commits">
+                <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a href="https://github.com/muaz-khan/WebRTC-Experiment/commits/master" target="_blank">Latest Updates</a></h2>
+                <div id="github-commits"></div>
+            </section>
         </article>
+        
+        <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel" class="fork-left"></a>
+        
         <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="http://twitter.com/muazkh" target="_blank">@muazkh</a>»
-                <a href="https://github.com/muaz-khan" target="_blank">Github</a>
+                <a href="https://www.webrtc-experiment.com/">WebRTC Experiments</a>
+                © <a href="https://plus.google.com/+MuazKhan" rel="author" target="_blank">Muaz Khan</a>
+                <a href="mailto:muazkh@gmail.com" target="_blank">muazkh@gmail.com</a>
             </p>
         </footer>
-        <script src="https://www.webrtc-experiment.com/common.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/DetectRTC/README.md b/DetectRTC/README.md
index 771b7769..e46d1164 100644
--- a/DetectRTC/README.md
+++ b/DetectRTC/README.md
@@ -3,14 +3,14 @@
 > A library for detecting WebRTC features!
 > Demo: https://www.webrtc-experiment.com/DetectRTC/
 
-<img src="https://www.webrtc-experiment.com/images/DetectRTC.png" style="width:100%;" />
+<img src="//cdn.webrtc-experiment.com/images/DetectRTC.png" style="width:100%;" />
 
 =
 
 ### How to use it?
 
 ```html
-<script src="//www.webrtc-experiment.com/DetectRTC.js"></script>
+<script src="//cdn.webrtc-experiment.com/DetectRTC.js"></script>
 ```
 
 =
diff --git a/DetectRTC/index.html b/DetectRTC/index.html
index d71c0258..4713fc46 100644
--- a/DetectRTC/index.html
+++ b/DetectRTC/index.html
@@ -13,7 +13,8 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio {
@@ -53,7 +54,7 @@
         </script>
         
         <!-- script used to detect WebRTC features! -->
-        <script src="https://www.webrtc-experiment.com/DetectRTC.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/DetectRTC.js"> </script>
     </head>
 
     <body>
@@ -85,7 +86,7 @@ <h1>
             <div class="github-stargazers"></div>
             
             <section class="experiment">
-                <img src="https://www.webrtc-experiment.com/images/DetectRTC.png" style="width:100%;" />
+                <img src="//cdn.webrtc-experiment.com/images/DetectRTC.png" style="width:100%;" />
                 <table id="browser-features"></table>
             </section>
             
@@ -119,7 +120,7 @@ <h2 class="header" id="feedback">Feedback</h2>
                 <h2 class="header">
                     How to use <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DetectRTC" target="_blank">DetectRTC</a>?</h2>
                 <pre>
-&lt;script src="https://www.WebRTC-Experiment.com/DetectRTC.js"&gt;&lt;/script&gt;
+&lt;script src="//cdn.WebRTC-Experiment.com/DetectRTC.js"&gt;&lt;/script&gt;
 </pre>
             </section>
             <section class="experiment">
@@ -179,6 +180,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/MultiRTC-simple/index.html b/MultiRTC-simple/index.html
index b7a0a980..ffe7db9a 100644
--- a/MultiRTC-simple/index.html
+++ b/MultiRTC-simple/index.html
@@ -21,7 +21,7 @@
         <link href="scrol-bars.css" rel="stylesheet">
         
         <!-- www.RTCMultiConnection.org/changes-log/ -->
-        <script src="//www.RTCMultiConnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
         
         <script src="linkify.js"> </script>
     </head>
diff --git a/Pluginfree-Screen-Sharing/README.md b/Pluginfree-Screen-Sharing/README.md
index 2dcd80fd..bfaff924 100644
--- a/Pluginfree-Screen-Sharing/README.md
+++ b/Pluginfree-Screen-Sharing/README.md
@@ -25,7 +25,7 @@ It isn't totally pluginfree (unfortunately)! You'll be asked either to enable co
     <li>
         Otherwise make sure you are running the chrome with command line flag "<strong><a href="http://peter.sh/experiments/chromium-command-line-switches/#enable-usermedia-screen-capturing">--enable-usermedia-screen-capturing</a></strong>" e.g. on Windows "<strong>Chrome.exe --enable-usermedia-screen-capturing</strong>"
         <div style="text-align:center;">
-            <img src="https://www.webrtc-experiment.com/images/--enable-usermedia-screen-capturing.png" style="max-width: 100%;">
+            <img src="//cdn.webrtc-experiment.com/images/--enable-usermedia-screen-capturing.png" style="max-width: 100%;">
         </div>
     </li>
 </ol>
diff --git a/Pluginfree-Screen-Sharing/index.html b/Pluginfree-Screen-Sharing/index.html
index b2ed1dc6..b63827b8 100644
--- a/Pluginfree-Screen-Sharing/index.html
+++ b/Pluginfree-Screen-Sharing/index.html
@@ -14,7 +14,7 @@
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
         
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             video {
@@ -73,8 +73,8 @@
         <link rel="chrome-webstore-item" href="https://chrome.google.com/webstore/detail/ajhifddimkapgcifgcodmmfdlknahffk">
         
         <!-- scripts used for screen-sharing -->
-        <script src='https://www.webrtc-experiment.com/firebase.js'> </script>
-        <script src="https://www.webrtc-experiment.com/Pluginfree-Screen-Sharing/conference.js"> </script>
+        <script src='//cdn.webrtc-experiment.com/firebase.js'> </script>
+        <script src="//cdn.webrtc-experiment.com/Pluginfree-Screen-Sharing/conference.js"> </script>
     </head>
 
     <body>
@@ -479,7 +479,7 @@ <h2>Prerequisites</h2>
                     <li>
                         Otherwise make sure you are running the chrome with command line flag "<strong><a href="http://peter.sh/experiments/chromium-command-line-switches/#enable-usermedia-screen-capturing">--enable-usermedia-screen-capturing</a></strong>" e.g. on Windows "<strong>Chrome.exe --enable-usermedia-screen-capturing</strong>"
                         <div style="text-align:center;">
-                            <img src="https://www.webrtc-experiment.com/images/--enable-usermedia-screen-capturing.png" style="max-width: 100%;">
+                            <img src="//cdn.webrtc-experiment.com/images/--enable-usermedia-screen-capturing.png" style="max-width: 100%;">
                         </div>
                         Learn more about <a href="http://www.chromium.org/developers/how-tos/run-chromium-with-flags" target="_blank">how to enable command-line flags</a> on different operating systems.
                     </li>
@@ -567,6 +567,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/README.md b/README.md
index 8d0fde18..5857360d 100644
--- a/README.md
+++ b/README.md
@@ -239,7 +239,7 @@
 ##### How to record audio using [RecordRTC](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC)?
 
 ```html
-<script src="//www.webrtc-experiment.com/RecordRTC.js"></script>
+<script src="//cdn.webrtc-experiment.com/RecordRTC.js"></script>
 ```
 
 ```javascript
@@ -274,7 +274,7 @@ You can write entire skype-like web-app using RTCMultiConnection! It supports al
 ```html
 <button id="openNewSessionButton">open New Session Button</button><br />
 
-<script src="http://www.RTCMultiConnection.org/latest.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
 <script>
 var connection = new RTCMultiConnection().connect();
 document.getElementById('openNewSessionButton').onclick = function() {
@@ -290,7 +290,7 @@ document.getElementById('openNewSessionButton').onclick = function() {
 ##### [DataChannel.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/DataChannel) / A library for RTCDataChannel APIs
 
 ```html
-<script src="//www.webrtc-experiment.com/DataChannel.js"> </script>
+<script src="//cdn.webrtc-experiment.com/DataChannel.js"> </script>
 <script>
     var channel = new DataChannel();
     channel.onopen = function(userid) {};
@@ -314,7 +314,7 @@ document.getElementById('openNewSessionButton').onclick = function() {
 Translator.js is a JavaScript library built top on Google Speech-Recognition & Translation API to transcript and translate voice and text. It supports many locales and brings globalization in <a href="https://www.webrtc-experiment.com/">WebRTC</a>!
 
 ```html
-<script src="//www.webrtc-experiment.com/Translator.js"> </script>
+<script src="//cdn.webrtc-experiment.com/Translator.js"> </script>
 ```
 
 ```javascript
diff --git a/RTCMultiConnection/demos/AppRTC-Look.html b/RTCMultiConnection/demos/AppRTC-Look.html
index b8068fca..56daae48 100644
--- a/RTCMultiConnection/demos/AppRTC-Look.html
+++ b/RTCMultiConnection/demos/AppRTC-Look.html
@@ -10,8 +10,8 @@
 <meta name="author" content="Muaz Khan">
 <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
 
-<script src="//www.rtcmulticonnection.org/latest.js"></script>
-<script src="//www.rtcmulticonnection.org/firebase.js"></script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"></script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"></script>
 
 <style>
 body, html {
diff --git a/RTCMultiConnection/demos/Multi-Broadcasters-and-Many-Viewers.html b/RTCMultiConnection/demos/Multi-Broadcasters-and-Many-Viewers.html
index c3244373..eb88e1c6 100644
--- a/RTCMultiConnection/demos/Multi-Broadcasters-and-Many-Viewers.html
+++ b/RTCMultiConnection/demos/Multi-Broadcasters-and-Many-Viewers.html
@@ -13,7 +13,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         <link rel="stylesheet" href="/style.css">
         <style>
             audio, video {
@@ -58,8 +58,8 @@
         </script>
         
         <!-- scripts used for WebRTC connections -->
-        <script src="//www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="//www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"></script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"></script>
     </head>
 
     <body>
@@ -211,6 +211,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/OneWay-Screen-TwoWay-Audio.html b/RTCMultiConnection/demos/OneWay-Screen-TwoWay-Audio.html
index 1985ecd5..74637820 100644
--- a/RTCMultiConnection/demos/OneWay-Screen-TwoWay-Audio.html
+++ b/RTCMultiConnection/demos/OneWay-Screen-TwoWay-Audio.html
@@ -6,7 +6,7 @@
 
 <title>OneWay Screen & Two-Way Audio using RTCMultiConnection</title>
 
-<link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+<link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
 <style>
     video {
         vertical-align: top;
@@ -20,8 +20,8 @@ <h1>OneWay Screen & Two-Way Audio using <a href="http://www.rtcmulticonnection.o
 <button id="share-screen-and-audio">Share Screen + Audio</button><hr />
 <div id="container"></div>
 
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
 
 <script>
     var connection = new RTCMultiConnection();
diff --git a/RTCMultiConnection/demos/RTCMultiConnection-v1.3-and-socket.io.html b/RTCMultiConnection/demos/RTCMultiConnection-v1.3-and-socket.io.html
index baf18a0d..2e43d674 100644
--- a/RTCMultiConnection/demos/RTCMultiConnection-v1.3-and-socket.io.html
+++ b/RTCMultiConnection/demos/RTCMultiConnection-v1.3-and-socket.io.html
@@ -4,8 +4,8 @@
 > Documentation - www.RTCMultiConnection.org
 -->
 
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
 
 <button id="init">Open NEW Session</button>
 
diff --git a/RTCMultiConnection/demos/RTCMultiConnection-v1.3-demo.html b/RTCMultiConnection/demos/RTCMultiConnection-v1.3-demo.html
index 962042cc..67f55696 100644
--- a/RTCMultiConnection/demos/RTCMultiConnection-v1.3-demo.html
+++ b/RTCMultiConnection/demos/RTCMultiConnection-v1.3-demo.html
@@ -22,8 +22,8 @@
                 location.reload();
             }
         </script>
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
     
     <body>
diff --git a/RTCMultiConnection/demos/RTCMultiConnection.sharePartOfScreen.html b/RTCMultiConnection/demos/RTCMultiConnection.sharePartOfScreen.html
index fa800edc..32cd99a1 100644
--- a/RTCMultiConnection/demos/RTCMultiConnection.sharePartOfScreen.html
+++ b/RTCMultiConnection/demos/RTCMultiConnection.sharePartOfScreen.html
@@ -95,8 +95,8 @@ <h3>Content is edit-able.</h3>
 
 
 <script src="//www.webrtc-experiment.com/screenshot.js"> </script>
-<script src="//www.rtcmulticonnection.org/latest.js"> </script>
-<script src="//www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"></script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"></script>
 
 <script>
 var connection = new RTCMultiConnection();
diff --git a/RTCMultiConnection/demos/RecordRTC-and-RTCMultiConnection.html b/RTCMultiConnection/demos/RecordRTC-and-RTCMultiConnection.html
index a8b35bfb..84756ed7 100644
--- a/RTCMultiConnection/demos/RecordRTC-and-RTCMultiConnection.html
+++ b/RTCMultiConnection/demos/RecordRTC-and-RTCMultiConnection.html
@@ -13,7 +13,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         <style>
             audio, video {
                 -moz-transition: all 1s ease;
@@ -81,9 +81,9 @@
         </script>
         
         <!-- scripts used for broadcasting -->
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-        <script src="https://www.webrtc-experiment.com/RecordRTC.min.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RecordRTC.js"> </script>
     </head>
 
     <body>
@@ -399,6 +399,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="//www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/Renegotiation.html b/RTCMultiConnection/demos/Renegotiation.html
index 11458b81..bb47b371 100644
--- a/RTCMultiConnection/demos/Renegotiation.html
+++ b/RTCMultiConnection/demos/Renegotiation.html
@@ -128,8 +128,8 @@
                 padding: .4em .8em;
             }
         </style>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
     </head>
     <body>
         <button id="new-session">
diff --git a/RTCMultiConnection/demos/admin-guest.html b/RTCMultiConnection/demos/admin-guest.html
index f55947e1..ad0344eb 100644
--- a/RTCMultiConnection/demos/admin-guest.html
+++ b/RTCMultiConnection/demos/admin-guest.html
@@ -158,9 +158,9 @@
                 width: 48%;
             }
         </style>
-        <script src="//www.webrtc-experiment.com/RTCMultiConnection-v1.6.js"> </script>
-        <script src="//www.webrtc-experiment.com/getMediaElement.min.js"> </script>
-        <script src="//www.webrtc-experiment.com/socket.io.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection-v1.6.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/getMediaElement.min.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/socket.io.js"> </script>
     </head>
     <body>
         <h1>
@@ -392,6 +392,6 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="//www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/all-in-one.html b/RTCMultiConnection/demos/all-in-one.html
index 61edec03..335abe72 100644
--- a/RTCMultiConnection/demos/all-in-one.html
+++ b/RTCMultiConnection/demos/all-in-one.html
@@ -13,7 +13,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio, video {
@@ -84,9 +84,9 @@
         </script>
         
         <!-- webrtc library used for streaming -->
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.webrtc-experiment.com/getMediaElement.min.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/getMediaElement.min.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -468,6 +468,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="//www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/audio-conferencing-data-sharing.html b/RTCMultiConnection/demos/audio-conferencing-data-sharing.html
index eae23860..e3093136 100644
--- a/RTCMultiConnection/demos/audio-conferencing-data-sharing.html
+++ b/RTCMultiConnection/demos/audio-conferencing-data-sharing.html
@@ -175,8 +175,8 @@
             document.createElement('footer');
         </script>
 
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection-v1.2.js"> </script>
     </head>
 
     <body>
@@ -373,5 +373,5 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
             </p>
         </footer>
     </body>
-    <script src="https://www.webrtc-experiment.com/common.js"> </script>
+    <script src="//cdn.webrtc-experiment.com/commits.js"> </script>
 </html>
diff --git a/RTCMultiConnection/demos/audioconferencing.html b/RTCMultiConnection/demos/audioconferencing.html
index 40d87608..30c2985c 100644
--- a/RTCMultiConnection/demos/audioconferencing.html
+++ b/RTCMultiConnection/demos/audioconferencing.html
@@ -13,7 +13,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio {
@@ -58,8 +58,8 @@
         </script>
         
         <!-- scripts used for broadcasting -->
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -184,6 +184,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/bandwidth.html b/RTCMultiConnection/demos/bandwidth.html
index b6bdf7df..461e21d0 100644
--- a/RTCMultiConnection/demos/bandwidth.html
+++ b/RTCMultiConnection/demos/bandwidth.html
@@ -214,8 +214,8 @@
             document.createElement('article');
             document.createElement('footer');
         </script>
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
     
     <body>
@@ -413,7 +413,7 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
 
             </p>
         </footer>
-        <script src="https://www.webrtc-experiment.com/common.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js"> </script>
     </body>
 
 </html>
diff --git a/RTCMultiConnection/demos/changeBandwidth.html b/RTCMultiConnection/demos/changeBandwidth.html
index d742131a..272d513e 100644
--- a/RTCMultiConnection/demos/changeBandwidth.html
+++ b/RTCMultiConnection/demos/changeBandwidth.html
@@ -1,5 +1,5 @@
-<script src="//www.rtcmulticonnection.org/latest.js"></script>
-<script src="//www.rtcmulticonnection.org/firebase.js"></script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"></script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"></script>
 
 <button id="openNewSessionButton">Open New Room</button>
 <button id="changeBandwidth" disabled>changeBandwidth</button><br />
diff --git a/RTCMultiConnection/demos/chrome-to-firefox-screen-sharing.html b/RTCMultiConnection/demos/chrome-to-firefox-screen-sharing.html
index f002e566..1877a350 100644
--- a/RTCMultiConnection/demos/chrome-to-firefox-screen-sharing.html
+++ b/RTCMultiConnection/demos/chrome-to-firefox-screen-sharing.html
@@ -9,8 +9,8 @@ <h1>Chrome-to-Firefox Screen Sharing Demo / using <a href="http://www.RTCMultiCo
 <button id="share-screen">Share Screen</button>
 <hr />
 <style>video { width: 80%; }</style>
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
 <script>
     var connection = new RTCMultiConnection();
     connection.session = {
@@ -20,7 +20,7 @@ <h1>Chrome-to-Firefox Screen Sharing Demo / using <a href="http://www.RTCMultiCo
 
     // forcing STUN; no TURN will be used!
     connection.iceServers = [{
-        url: !moz ? 'stun:stun.l.google.com:19302' : 'stun:23.21.150.121'
+        url: 'stun:stun.l.google.com:19302'
     }];
 
     // forcing "OfferToReceiveAudio:false"
diff --git a/RTCMultiConnection/demos/features.html b/RTCMultiConnection/demos/features.html
index b4564f1d..f9e3bba4 100644
--- a/RTCMultiConnection/demos/features.html
+++ b/RTCMultiConnection/demos/features.html
@@ -49,8 +49,8 @@ <h1><a href="http://www.RTCMultiConnection.org/docs/">RTCMultiConnection</a> Fea
 </select>
 <hr />
 
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
 
 <script>
     var connection;
diff --git a/RTCMultiConnection/demos/getMediaDevices.html b/RTCMultiConnection/demos/getMediaDevices.html
index 3c789871..586cf0ac 100644
--- a/RTCMultiConnection/demos/getMediaDevices.html
+++ b/RTCMultiConnection/demos/getMediaDevices.html
@@ -14,9 +14,8 @@
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
         
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
-        <link rel="stylesheet" href="/style.css">
         <style>
             audio, video {
                 -moz-transition: all 1s ease;
@@ -59,7 +58,8 @@
             document.createElement('footer');
         </script>
         
-        <script src="//www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"></script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"></script>
     </head>
 
     <body>
@@ -180,6 +180,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/group-file-sharing-plus-text-chat.html b/RTCMultiConnection/demos/group-file-sharing-plus-text-chat.html
index 637afe29..99ab310a 100644
--- a/RTCMultiConnection/demos/group-file-sharing-plus-text-chat.html
+++ b/RTCMultiConnection/demos/group-file-sharing-plus-text-chat.html
@@ -164,8 +164,8 @@
             document.createElement('footer');
         </script>
 
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -394,6 +394,6 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
             </p>
         </footer>
-        <script src="https://www.webrtc-experiment.com/common.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js"> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/join-with-or-without-camera.html b/RTCMultiConnection/demos/join-with-or-without-camera.html
index 0eeb7f7a..b931bc3e 100644
--- a/RTCMultiConnection/demos/join-with-or-without-camera.html
+++ b/RTCMultiConnection/demos/join-with-or-without-camera.html
@@ -193,8 +193,8 @@
             document.createElement('footer');
         </script>
 
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -344,6 +344,6 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
             </p>
         </footer>
-        <script src="https://www.webrtc-experiment.com/common.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js"> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html b/RTCMultiConnection/demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html
index c20a448c..d0eb6b75 100644
--- a/RTCMultiConnection/demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html
+++ b/RTCMultiConnection/demos/manual-session-establishment-plus-extra-data-transmission-plus-videoconferencing.html
@@ -182,8 +182,8 @@
             document.createElement('footer');
         </script>
 
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection-v1.2.js"> </script>
     </head>
 
     <body>
@@ -395,6 +395,6 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
             </p>
         </footer>
-        <script src="https://www.webrtc-experiment.com/common.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js"> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/manual-session-establishment-plus-extra-data-transmission.html b/RTCMultiConnection/demos/manual-session-establishment-plus-extra-data-transmission.html
index 1b584cc7..e344839b 100644
--- a/RTCMultiConnection/demos/manual-session-establishment-plus-extra-data-transmission.html
+++ b/RTCMultiConnection/demos/manual-session-establishment-plus-extra-data-transmission.html
@@ -164,8 +164,8 @@
             document.createElement('footer');
         </script>
 
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.2.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection-v1.2.js"> </script>
     </head>
 
     <body>
@@ -412,6 +412,6 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
                 <a href="https://github.com/muaz-khan" target="_blank">Github</a>
             </p>
         </footer>
-        <script src="https://www.webrtc-experiment.com/common.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js"> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/multi-session-establishment.html b/RTCMultiConnection/demos/multi-session-establishment.html
index 4f4ee308..1e8dfea0 100644
--- a/RTCMultiConnection/demos/multi-session-establishment.html
+++ b/RTCMultiConnection/demos/multi-session-establishment.html
@@ -15,7 +15,7 @@
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
         
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio, video {
@@ -52,8 +52,8 @@
             document.createElement('footer');
         </script>
         
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -230,7 +230,7 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 
 </html>
diff --git a/RTCMultiConnection/demos/multi-sessions-management.html b/RTCMultiConnection/demos/multi-sessions-management.html
index 18e34d50..775bfa32 100644
--- a/RTCMultiConnection/demos/multi-sessions-management.html
+++ b/RTCMultiConnection/demos/multi-sessions-management.html
@@ -15,8 +15,8 @@ <h1><a href="http://www.RTCMultiConnection.org/docs/">RTCMultiConnection</a> & M
 <hr />
 
 
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
 <script>
     document.querySelector('#open-connections').onclick = function() {
         audioVideoConnection.open();
diff --git a/RTCMultiConnection/demos/multi-streams-attachment.html b/RTCMultiConnection/demos/multi-streams-attachment.html
index 2a0d4c0e..1832b46e 100644
--- a/RTCMultiConnection/demos/multi-streams-attachment.html
+++ b/RTCMultiConnection/demos/multi-streams-attachment.html
@@ -12,8 +12,8 @@ <h1>
         width: 48%;
     }
 </style>
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
 <button id="init">
     Open New Connection</button>
 <br />
diff --git a/RTCMultiConnection/demos/mute-unmute.html b/RTCMultiConnection/demos/mute-unmute.html
index 62032bc8..49aa1461 100644
--- a/RTCMultiConnection/demos/mute-unmute.html
+++ b/RTCMultiConnection/demos/mute-unmute.html
@@ -6,8 +6,8 @@
 <h1><a href="http://www.RTCMultiConnection.org/docs/">RTCMultiConnection</a> & Mute/UnMute</h1>
 <button id="setup">Setup RTC-Multi-Connection</button><button id="mute">Mute</button><hr />
 
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
 <script>
     document.querySelector('#setup').onclick = function() {
         connection.open();
diff --git a/RTCMultiConnection/demos/one-page-demo.html b/RTCMultiConnection/demos/one-page-demo.html
index 2422b55e..cb5c7c92 100644
--- a/RTCMultiConnection/demos/one-page-demo.html
+++ b/RTCMultiConnection/demos/one-page-demo.html
@@ -14,7 +14,7 @@
 
 <h1><a href="http://www.RTCMultiConnection.org/docs/">RTCMultiConnection</a> one-page demo</h1>
 
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
 <button>Setup RTC-Multi-Connection</button><hr />
 
 <script>
diff --git a/RTCMultiConnection/demos/one-to-one-filesharing.html b/RTCMultiConnection/demos/one-to-one-filesharing.html
index 20df3672..bcdad39b 100644
--- a/RTCMultiConnection/demos/one-to-one-filesharing.html
+++ b/RTCMultiConnection/demos/one-to-one-filesharing.html
@@ -15,15 +15,15 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         <!-- for HTML5 el styling -->
         <script>
             document.createElement('article');
             document.createElement('footer');
         </script>
 
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -132,6 +132,6 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/password-protect-rooms.html b/RTCMultiConnection/demos/password-protect-rooms.html
index 2a9ad07e..2a7fa0f3 100644
--- a/RTCMultiConnection/demos/password-protect-rooms.html
+++ b/RTCMultiConnection/demos/password-protect-rooms.html
@@ -6,8 +6,8 @@
 <h1><a href="http://www.RTCMultiConnection.org/docs/">RTCMultiConnection</a> & Password Protect Rooms</h1>
 <button id="setup">Setup RTC-Multi-Connection</button><hr />
 
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
 <script>
     document.querySelector('#setup').onclick = function() {
         // room password has been set before calling "open" method
diff --git a/RTCMultiConnection/demos/remote-stream-forwarding.html b/RTCMultiConnection/demos/remote-stream-forwarding.html
index ebc92fcf..2c286b9e 100644
--- a/RTCMultiConnection/demos/remote-stream-forwarding.html
+++ b/RTCMultiConnection/demos/remote-stream-forwarding.html
@@ -15,7 +15,7 @@
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
         
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio, video {
@@ -59,8 +59,8 @@
             document.createElement('footer');
         </script>
         
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
     </head>
 
     <body>
@@ -237,6 +237,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
         
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/rooms-screenshots.html b/RTCMultiConnection/demos/rooms-screenshots.html
index 2d507231..7a01cb50 100644
--- a/RTCMultiConnection/demos/rooms-screenshots.html
+++ b/RTCMultiConnection/demos/rooms-screenshots.html
@@ -50,8 +50,8 @@
     }
 </style>
 
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
 
 <h1>Preview Screenshot of the room using <a href="http://www.RTCMultiConnection.org/docs/">RTCMultiConnection.js</a></h1>
 
@@ -76,8 +76,8 @@ <h1>Preview Screenshot of the room using <a href="http://www.RTCMultiConnection.
 <hr />
 <script>
     var images = [
-        '//www.webrtc-experiment.com/images/javascript.png',
-        '//www.webrtc-experiment.com/images/ajax-loader.gif'
+        '//cdn.webrtc-experiment.com/images/javascript.png',
+        '//cdn.webrtc-experiment.com/images/ajax-loader.gif'
     ];
 
     var connection = new RTCMultiConnection();
diff --git a/RTCMultiConnection/demos/screen-sharing.html b/RTCMultiConnection/demos/screen-sharing.html
index 6cd9ada8..af5980e3 100644
--- a/RTCMultiConnection/demos/screen-sharing.html
+++ b/RTCMultiConnection/demos/screen-sharing.html
@@ -13,7 +13,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio, video {
@@ -59,8 +59,8 @@
         </script>
         
         <!-- scripts used for broadcasting -->
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -246,6 +246,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/session-management.html b/RTCMultiConnection/demos/session-management.html
index f2f4f0cd..c6f3d694 100644
--- a/RTCMultiConnection/demos/session-management.html
+++ b/RTCMultiConnection/demos/session-management.html
@@ -6,8 +6,8 @@
 
 <h1><a href="http://www.RTCMultiConnection.org/docs/">RTCMultiConnection</a> & Session Management</h1>
 
-<script src="https://www.rtcmulticonnection.org/latest.js"> </script>
-<script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
 
 <button id="share-audio-video" class="button">Audio+Video</button>
 <button id="share-screen" class="button">Screen</button>
diff --git a/RTCMultiConnection/demos/session-reinitiation.html b/RTCMultiConnection/demos/session-reinitiation.html
index 055032f0..839d8472 100644
--- a/RTCMultiConnection/demos/session-reinitiation.html
+++ b/RTCMultiConnection/demos/session-reinitiation.html
@@ -13,7 +13,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         <style>
             audio, video {
                 -moz-transition: all 1s ease;
@@ -57,8 +57,8 @@
         </script>
         
         <!-- scripts used for broadcasting -->
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -275,6 +275,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/stream-mp3-live.html b/RTCMultiConnection/demos/stream-mp3-live.html
index c9189140..5b0aa061 100644
--- a/RTCMultiConnection/demos/stream-mp3-live.html
+++ b/RTCMultiConnection/demos/stream-mp3-live.html
@@ -1,5 +1,5 @@
-<script src="//www.rtcmulticonnection.org/latest.js"></script>
-<script src="//www.rtcmulticonnection.org/firebase.js"></script>
+<script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"></script>
+<script src="//cdn.webrtc-experiment.com/firebase.js"></script>
 
 <input type="file">
 <button id="openNewSessionButton" disabled>Open New Room</button><br />
diff --git a/RTCMultiConnection/demos/users-ejection.html b/RTCMultiConnection/demos/users-ejection.html
index 8cb1c84c..49f4e2cf 100644
--- a/RTCMultiConnection/demos/users-ejection.html
+++ b/RTCMultiConnection/demos/users-ejection.html
@@ -14,7 +14,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
 
         <style>
             audio, video {
@@ -90,8 +90,8 @@
             document.createElement('footer');
         </script>
 
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -277,6 +277,6 @@ <h2 id="feedback" style="border-bottom: 1px solid rgb(189, 189, 189); padding: .
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/video-broadcasting.html b/RTCMultiConnection/demos/video-broadcasting.html
index e21ee6aa..de3aafd8 100644
--- a/RTCMultiConnection/demos/video-broadcasting.html
+++ b/RTCMultiConnection/demos/video-broadcasting.html
@@ -13,7 +13,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio, video {
@@ -59,8 +59,8 @@
         </script>
         
         <!-- scripts used for broadcasting -->
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -237,6 +237,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RTCMultiConnection/demos/videoconferencing.html b/RTCMultiConnection/demos/videoconferencing.html
index 914b33eb..35268429 100644
--- a/RTCMultiConnection/demos/videoconferencing.html
+++ b/RTCMultiConnection/demos/videoconferencing.html
@@ -13,7 +13,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio, video {
@@ -58,8 +58,8 @@
         </script>
         
         <!-- scripts used for broadcasting -->
-        <script src="https://www.rtcmulticonnection.org/firebase.js"> </script>
-        <script src="https://www.rtcmulticonnection.org/latest.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCMultiConnection.js"> </script>
     </head>
 
     <body>
@@ -238,6 +238,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RecordRTC-Development/README.md b/RecordRTC-Development/README.md
deleted file mode 100644
index fd90e7ca..00000000
--- a/RecordRTC-Development/README.md
+++ /dev/null
@@ -1,377 +0,0 @@
-## [RecordRTC Development](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC-Development)
-
-For early [RecordRTC](https://www.webrtc-experiment.com/RecordRTC/) experiments.
-
-New fixes:
-
-1. Longer video recordings on Chrome!
-2. Advertisement option for video recordings!
-
-=
-
-#### RecordRTC: WebRTC audio/video recording / [Demo](https://www.webrtc-experiment.com/RecordRTC/)
-
-[RecordRTC](https://www.webrtc-experiment.com/RecordRTC/) is a server-less (entire client-side) JavaScript library can be used to record WebRTC audio/video media streams. It supports cross-browser audio/video recording.
-
-=
-
-1. [RecordRTC to Node.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/RecordRTC-to-Nodejs)
-2. [RecordRTC to PHP](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/RecordRTC-to-PHP)
-3. [RecordRTC to ASP.NET MVC](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/RecordRTC-to-ASPNETMVC)
-4. [RecordRTC & HTML-2-Canvas i.e. Canvas/HTML Recording!](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/Canvas-Recording)
-5. [MRecordRTC i.e. Multi-RecordRTC!](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/MRecordRTC)
-6. [RecordRTC on Ruby!](https://github.com/cbetta/record-rtc-experiment)
-7. [RecordRTC over Socket.io](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/RecordRTC-over-Socketio)
-8. [ffmpeg-asm.js and RecordRTC! Audio/Video Merging & Transcoding!](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/ffmpeg)
-
-=
-
-```html
-<script src="//www.WebRTC-Experiment.com/RecordRTC.js"></script>
-```
-
-=
-
-#### How to record audio?
-
-```javascript
-var recordRTC = RecordRTC(mediaStream);
-recordRTC.startRecording();
-recordRTC.stopRecording(function(audioURL) {
-   mediaElement.src = audioURL;
-});
-```
-
-Remember, you need to invoke `navigator.getUserMedia` method yourself; it is too easy to use!
-
-```javascript
-navigator.getUserMedia({audio: true}, function(mediaStream) {
-   window.recordRTC = RecordRTC(MediaStream);
-   recordRTC.startRecording();
-});
-
-btnStopRecording.onclick = function() {
-   recordRTC.stopRecording(function(audioURL) {
-      window.open(audioURL);
-   });
-};
-```
-
-Also, you don't need to use prefixed versions of `getUserMedia` and `URL` objects. RecordRTC auto handles such things for you! Just use non-prefixed version:
-
-```javascript
-navigator.getUserMedia(media_constraints, onsuccess, onfailure);
-URL.createObjectURL(MediaStream);
-```
-
-=
-
-#### How to record video?
-
-Everything is optional except `type:'video'`:
-
-```javascript
-var options = {
-   type: 'video'
-};
-var recordRTC = RecordRTC(mediaStream, options);
-recordRTC.startRecording();
-recordRTC.stopRecording(function(videoURL) {
-   mediaElement.src = videoURL;
-});
-```
-
-=
-
-##### How to record animated GIF image?
-
-Everything is optional except `type:'gif'`:
-
-```javascript
-var options = {
-   type: 'gif',
-   frameRate: 200,
-   quality: 10
-};
-var recordRTC = RecordRTC(mediaStream, options);
-recordRTC.startRecording();
-recordRTC.stopRecording(function(gifURL) {
-   mediaElement.src = gifURL;
-});
-```
-
-=
-
-##### How to record HTML2Canvas?
-
-You can say it: "HTML/Canvas Recording using RecordRTC"!
-
-```html
-<script src="//www.WebRTC-Experiment.com/RecordRTC.js"></script>
-<script src="//www.webrtc-experiment.com/screenshot.js"></script>
-<div id="elementToShare" style="width:100%;height:100%;background:green;"></div>
-<script>
-var elementToShare = document.getElementById('elementToShare');
-var recordRTC = RecordRTC(elementToShare, {
-    type: 'canvas'
-});
-recordRTC.startRecording();
-recordRTC.stopRecording(function(url) {
-    window.open(url);
-});
-</script>
-```
-
-See a demo: https://www.webrtc-experiment.com/RecordRTC/Canvas-Recording/
-
-=
-
-##### `autoWriteToDisk`
-
-Using `autoWriteToDisk`; you can suggest RecordRTC to auto-write to indexed-db as soon as you call `stopRecording` method.
-
-```javascript
-var recordRTC = RecordRTC(MediaStream, {
-    autoWriteToDisk: true
-});
-```
-
-`autoWriteToDisk` is helpful for single stream recording and writing to disk; however for `MRecordRTC`; `writeToDisk` is preferred one.
-
-=
-
-##### `writeToDisk`
-
-You can write recorded blob to disk using `writeToDisk` method:
-
-```javascript
-recordRTC.stopRecording();
-recordRTC.writeToDisk();
-```
-
-=
-
-##### `getFromDisk`
-
-You can get recorded blob from disk using `getFromDisk` method:
-
-```javascript
-// get all blobs from disk
-RecordRTC.getFromDisk('all', function(dataURL, type) {
-   type == 'audio'
-   type == 'video'
-   type == 'gif'
-});
-
-// or get just single blob
-RecordRTC.getFromDisk('audio', function(dataURL) {
-   // only audio blob is returned from disk!
-});
-```
-
-For [MRecordRTC](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/MRecordRTC); you can use word `MRecordRTC` instead of `RecordRTC`!
-
-Another possible situation!
-
-```javascript
-var recordRTC = RecordRTC(mediaStream);
-recordRTC.startRecording();
-recordRTC.stopRecording(function(audioURL) {
-   mediaElement.src = audioURL;
-});
-
-// "recordRTC" instance object to invoke "getFromDisk" method!
-recordRTC.getFromDisk(function(dataURL) {
-   // audio blob is automaticlaly returned from disk!
-});
-```
-
-In the above example; you can see that `recordRTC` instance object is used instead of global `RecordRTC` object.
-
-=
-
-##### How to set video width/height?
-
-```javascript
-var options = {
-   type: 'video',
-   video: {
-      width: 320,
-      height: 240
-   },
-   canvas: {
-      width: 320,
-      height: 240
-   }
-};
-```
-
-=
-
-##### How to get DataURL?
-
-```javascript
-recordRTC.getDataURL(function(dataURL) {
-   mediaElement.src = dataURL;
-});
-```
-
-=
-
-##### How to get `Blob` object?
-
-```javascript
-blob = recordRTC.getBlob();
-```
-
-=
-
-##### How to get Virtual-URL?
-
-```javascript
-window.open( recordRTC.toURL() );
-```
-
-=
-
-##### How to invoke save-to-disk dialog?
-
-```javascript
-recordRTC.save();
-```
-
-=
-
-##### How to customize Buffer-Size for audio recording?
-
-```javascript
-var options = {
-   bufferSize: 16384
-};
-var recordRTC = RecordRTC(audioStream, options);
-```
-
-Following values are allowed for buffer-size:
-
-```javascript
-// Legal values are (256, 512, 1024, 2048, 4096, 8192, 16384)
-```
-
-You can write like this:
-
-```javascript
-var options = {
-   'buffer-size': 16384
-};
-```
-
-=
-
-##### How to customize Sample-Rate for audio recording?
-
-```javascript
-var options = {
-   sampleRate: 96000
-};
-var recordRTC = RecordRTC(audioStream, options);
-```
-
-Values for sample-rate must be greater than or equal to 22050 and less than or equal to 96000.
-
-You can write like this:
-
-```javascript
-var options = {
-   'sample-rate': 16384
-};
-```
-
-=
-
-##### Is WinXP supported?
-
-No WinXP SP2 support. However, RecordRTC works on WinXP Service Pack 3.
-
-=
-
-##### Is Chrome on Android supported?
-
-RecordRTC uses WebAudio API for stereo-audio recording. AFAIK, WebAudio is not supported on android chrome releases, yet.
-
-=
-
-##### Stereo or Mono?
-
-Audio recording fails for `mono` audio. So, try `stereo` audio only.
-
-=
-
-##### Possible issues/failures:
-
-Do you know "RecordRTC" fails recording audio because following conditions fails:
-
-1. Sample rate and channel configuration must be the same for input and output sides on Windows i.e. audio input/output devices mismatch
-2. Only the Default microphone device can be used for capturing.
-3. The requesting scheme is none of the following: http, https, chrome, extension's, or file (only works with `--allow-file-access-from-files`)
-4. The browser cannot create/initialize the metadata database for the API under the profile directory
-
-If you see this error message: `Uncaught Error: SecurityError: DOM Exception 18`; it means that you're using `HTTP`; whilst your webpage is loading worker file (i.e. `audio-recorder.js`) from `HTTPS`. Both files's (i.e. `RecordRTC.js` and `audio-recorder.js`) scheme MUST be same!
-
-=
-
-##### Web Audio APIs requirements
-
-1. If you're on Windows, you have to be running WinXP SP3, Windows Vista or better (will not work on Windows XP SP2 or earlier).
-2. On Windows, audio input hardware must be set to the same sample rate as audio output hardware.
-3. On Mac and Windows, the audio input device must be at least stereo (i.e. a mono/single-channel USB microphone WILL NOT work).
-
-=
-
-##### Why stereo?
-
-If you explorer chromium code; you'll see that some APIs can only be successfully called for `WAV` files with `stereo` audio.
-
-Stereo audio is only supported for WAV files.
-
-...still investigating the actual issue of failure with `mono` audio.
-
-=
-
-Media Stream Recording API (MediaRecorder object) is being implemented by both Firefox and Chrome. RecordRTC is also using MediaRecorder API for Firefox (nightly).
-
-RecordRTC is unable to record "mono" audio on chrome; however it seems that we can covert channels from "stereo" to "mono" using WebAudio API, though. MediaRecorder API's encoder only support 48k/16k mono audio channel (on Firefox Nightly).
-
-=
-
-##### Browser Support
-
-[RecordRTC Demo](https://www.webrtc-experiment.com/RecordRTC/) works fine on following web-browsers:
-
-| Browser        | Support           |
-| ------------- |-------------|
-| 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) |
-| Firefox | [Nightly](http://nightly.mozilla.org/) |
-
-=
-
-##### Credits
-
-1. [Recorderjs](https://github.com/mattdiamond/Recorderjs) for audio recording
-2. [whammy](https://github.com/antimatter15/whammy) for video recording
-3. [jsGif](https://github.com/antimatter15/jsgif) for gif recording
-
-=
-
-##### Spec & Reference
-
-1. [Web Audio API](https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html)
-2. [MediaRecorder](https://wiki.mozilla.org/Gecko:MediaRecorder)
-3. [Canvas2D](http://www.w3.org/html/wg/drafts/2dcontext/html5_canvas/)
-4. [MediaStream Recording](https://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/MediaRecorder.html)
-5. [Media Capture and Streams](http://www.w3.org/TR/mediacapture-streams/)
-
-=
-
-## License
-
-[RecordRTC.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC) is released under [MIT licence](https://www.webrtc-experiment.com/licence/) . Copyright (c) [Muaz Khan](https://plus.google.com/+MuazKhan).
diff --git a/RecordRTC-Development/RecordRTC.js b/RecordRTC-Development/RecordRTC.js
deleted file mode 100644
index 78ee354f..00000000
--- a/RecordRTC-Development/RecordRTC.js
+++ /dev/null
@@ -1,1520 +0,0 @@
-// Last time updated at June 14, 2014, 08:32:23
-
-//------------------------------------
-/*
--. breaking updates:
--. now, you MUST always set config-type == 'video' for Firefox; otherwise
--. RecordRTC will record audio/ogg instead of video/webm.
--. It doesn't matter which kind of stream you're passing. Multi-tracks or single track.
-*/
-
-// issues?
-// -. (fixed) audio recording while passing multi-tracks media stream for Firefox.
-// -. audio self-playback (ehco/noise/etc.)
-// -. need to fix: recordRTC.setAdvertisementArray( [ 'data:image-webp', 'data:image-webp', 'data:image-webp' ] );
-// -. it seems that RecordRTC is cutting off the last couple of seconds of recordings
-// -. ffmpeg-asm.js demos should encode in Hight Quality audio and video containers (ref issue #204)
-// -. 14K ogg is not a good output; it should be 128k ogg. VLC is unable to play it; it must!
-//------------------------------------
-
-// Browsers Support::
-// Chrome (all versions) [ audio/video individually ]
-// Firefox ( >= 29 ) [ audio/video in single webm/mp4 container or only audio in ogg ]
-// Opera (all versions) [ same as chrome ]
-// Android (Chrome) [ only video ]
-// Android (Opera) [ only video ]
-// Android (Firefox) [ only video ]
-
-//------------------------------------
-// Muaz Khan         - www.MuazKhan.com
-// MIT License       - www.WebRTC-Experiment.com/licence
-// Documentation     - github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC
-//------------------------------------
-// Note: RecordRTC.js is using 3 other libraries; you need to accept their licences as well.
-//------------------------------------
-// 1. RecordRTC.js
-// 2. MRecordRTC.js
-// 3. Cross-Browser-Declarations.js
-// 4. Storage.js
-// 5. MediaStreamRecorder.js
-// 6. StereoRecorder.js
-// 7. StereoAudioRecorder.js
-// 8. CanvasRecorder.js
-// 9. WhammyRecorder.js
-// 10. Whammy.js
-// 11. DiskStorage.js
-// 12. GifRecorder.js
-// 13. gifEncoder.js
-//------------------------------------
-
-// ____________
-// RecordRTC.js
-
-function RecordRTC(mediaStream, config) {
-    config = config || {};
-
-    if (!mediaStream) throw 'MediaStream is mandatory.';
-    if (!config.type) config.type = 'audio';
-
-    function startRecording() {
-        console.debug('started recording ' + config.type + ' stream.');
-
-        // Media Stream Recording API has not been implemented in chrome yet;
-        // That's why using WebAudio API to record stereo audio in WAV format
-        var Recorder = IsChrome ? window.StereoRecorder : window.MediaStreamRecorder;
-
-        // video recorder (in WebM format)
-        if (config.type == 'video' && IsChrome) Recorder = window.WhammyRecorder;
-
-        // video recorder (in Gif format)
-        if (config.type == 'gif') Recorder = window.GifRecorder;
-
-        // html2canvas recording!
-        if (config.type == 'canvas') Recorder = window.CanvasRecorder;
-
-        mediaRecorder = new Recorder(mediaStream);
-
-        // Merge all data-types except "function"
-        mediaRecorder = mergeProps(mediaRecorder, config);
-        mediaRecorder.onAudioProcessStarted = function () {
-            if (config.onAudioProcessStarted) config.onAudioProcessStarted();
-        };
-
-        mediaRecorder.record();
-
-        return this;
-    }
-
-    function stopRecording(callback) {
-        if (!mediaRecorder) return console.warn(WARNING);
-
-        console.warn('stopped recording ' + config.type + ' stream.');
-
-        if ((config.type == 'audio') || (config.type == 'video') || (config.type == 'canvas')) {
-            mediaRecorder.stop(_callback);
-        } else {
-            mediaRecorder.stop();
-            _callback();
-        }
-
-        function _callback() {
-            var blob = mediaRecorder.recordedBlob;
-            if (callback) {
-                var url = URL.createObjectURL(blob);
-                callback(url);
-            }
-            console.debug(blob.type, '->', bytesToSize(blob.size));
-
-            if (config.autoWriteToDisk) {
-                getDataURL(function (dataURL) {
-                    var parameter = {};
-                    parameter[config.type + 'Blob'] = dataURL;
-                    DiskStorage.Store(parameter);
-                });
-            }
-        }
-    }
-
-    function getDataURL(callback, _mediaRecorder) {
-        if (!callback) throw 'Pass a callback function over getDataURL.';
-
-        _getDataURL();
-
-        function _getDataURL() {
-            if (!!window.Worker) {
-                var webWorker = processInWebWorker(function readFile(_blob) {
-                    postMessage(new FileReaderSync().readAsDataURL(_blob));
-                });
-
-                webWorker.onmessage = function (event) {
-                    callback(event.data);
-                };
-
-                webWorker.postMessage(_mediaRecorder ? _mediaRecorder.recordedBlob : mediaRecorder.recordedBlob);
-            } else {
-                var reader = new FileReader();
-                reader.readAsDataURL(_mediaRecorder ? _mediaRecorder.recordedBlob : mediaRecorder.recordedBlob);
-                reader.onload = function (event) {
-                    callback(event.target.result);
-                };
-            }
-        }
-
-        function processInWebWorker(_function) {
-            var blob = URL.createObjectURL(new Blob([_function.toString(),
-                'this.onmessage =  function (e) {readFile(e.data);}'
-            ], {
-                type: 'application/javascript'
-            }));
-
-            var worker = new Worker(blob);
-            URL.revokeObjectURL(blob);
-            return worker;
-        }
-    }
-
-    var WARNING = 'It seems that "startRecording" is not invoked for ' + config.type + ' recorder.';
-
-    var mediaRecorder;
-
-    return {
-        startRecording: startRecording,
-        stopRecording: stopRecording,
-        getBlob: function () {
-            if (!mediaRecorder) return console.warn(WARNING);
-            return mediaRecorder.recordedBlob;
-        },
-        getDataURL: getDataURL,
-        toURL: function () {
-            if (!mediaRecorder) return console.warn(WARNING);
-            return URL.createObjectURL(mediaRecorder.recordedBlob);
-        },
-        save: function () {
-            if (!mediaRecorder) return console.warn(WARNING);
-
-            // bug: should we use "getBlob" instead; to handle aww-snaps!
-            this.getDataURL(function (dataURL) {
-                var hyperlink = document.createElement('a');
-                hyperlink.href = dataURL;
-                hyperlink.target = '_blank';
-                hyperlink.download = (Math.round(Math.random() * 9999999999) + 888888888) + '.' + mediaRecorder.recordedBlob.type.split('/')[1];
-
-                var evt = new MouseEvent('click', {
-                    view: window,
-                    bubbles: true,
-                    cancelable: true
-                });
-
-                hyperlink.dispatchEvent(evt);
-
-                (window.URL || window.webkitURL).revokeObjectURL(hyperlink.href);
-            });
-        },
-        getFromDisk: function (callback) {
-            if (!mediaRecorder) return console.warn(WARNING);
-            RecordRTC.getFromDisk(config.type, callback);
-        },
-        setAdvertisementArray: function (arrayOfWebPImages) {
-            this.advertisement = [];
-
-            var length = arrayOfWebPImages.length;
-            for (var i = 0; i < length; i++) {
-                this.advertisement.push({
-                    duration: i,
-                    image: arrayOfWebPImages[i]
-                });
-            }
-        }
-    };
-}
-
-RecordRTC.getFromDisk = function (type, callback) {
-    if (!callback) throw 'callback is mandatory.';
-
-    console.log('Getting recorded ' + (type == 'all' ? 'blobs' : type + ' blob ') + ' from disk!');
-    DiskStorage.Fetch(function (dataURL, _type) {
-        if (type != 'all' && _type == type + 'Blob') {
-            if (callback) callback(dataURL);
-        }
-
-        if (type == 'all') {
-            if (callback) callback(dataURL, _type.replace('Blob', ''));
-        }
-    });
-};
-
-RecordRTC.writeToDisk = function (options) {
-    console.log('Writing recorded blob(s) to disk!');
-    options = options || {};
-    if (options.audio && options.video && options.gif) {
-        options.audio.getDataURL(function (audioDataURL) {
-            options.video.getDataURL(function (videoDataURL) {
-                options.gif.getDataURL(function (gifDataURL) {
-                    DiskStorage.Store({
-                        audioBlob: audioDataURL,
-                        videoBlob: videoDataURL,
-                        gifBlob: gifDataURL
-                    });
-                });
-            });
-        });
-    } else if (options.audio && options.video) {
-        options.audio.getDataURL(function (audioDataURL) {
-            options.video.getDataURL(function (videoDataURL) {
-                DiskStorage.Store({
-                    audioBlob: audioDataURL,
-                    videoBlob: videoDataURL
-                });
-            });
-        });
-    } else if (options.audio && options.gif) {
-        options.audio.getDataURL(function (audioDataURL) {
-            options.gif.getDataURL(function (gifDataURL) {
-                DiskStorage.Store({
-                    audioBlob: audioDataURL,
-                    gifBlob: gifDataURL
-                });
-            });
-        });
-    } else if (options.video && options.gif) {
-        options.video.getDataURL(function (videoDataURL) {
-            options.gif.getDataURL(function (gifDataURL) {
-                DiskStorage.Store({
-                    videoBlob: videoDataURL,
-                    gifBlob: gifDataURL
-                });
-            });
-        });
-    } else if (options.audio) {
-        options.audio.getDataURL(function (audioDataURL) {
-            DiskStorage.Store({
-                audioBlob: audioDataURL
-            });
-        });
-    } else if (options.video) {
-        options.video.getDataURL(function (videoDataURL) {
-            DiskStorage.Store({
-                videoBlob: videoDataURL
-            });
-        });
-    } else if (options.gif) {
-        options.gif.getDataURL(function (gifDataURL) {
-            DiskStorage.Store({
-                gifBlob: gifDataURL
-            });
-        });
-    }
-};
-
-// _____________
-// MRecordRTC.js
-
-function MRecordRTC(mediaStream) {
-    this.addStream = function (_mediaStream) {
-        if (_mediaStream) mediaStream = _mediaStream;
-    };
-
-    this.mediaType = {
-        audio: true,
-        video: true
-    };
-
-    this.startRecording = function () {
-        if (this.mediaType.audio) {
-            this.audioRecorder = RecordRTC(mediaStream, this).startRecording();
-        }
-
-        if (this.mediaType.video) {
-            this.videoRecorder = RecordRTC(mediaStream, {
-                type: 'video'
-            }).startRecording();
-        }
-
-        if (this.mediaType.gif) {
-            this.gifRecorder = RecordRTC(mediaStream, {
-                type: 'gif',
-                frameRate: this.frameRate || 200,
-                quality: this.quality || 10
-            }).startRecording();
-        }
-    };
-
-    this.stopRecording = function (callback) {
-        callback = callback || function () {};
-
-        if (this.audioRecorder) {
-            this.audioRecorder.stopRecording(function (blobURL) {
-                callback(blobURL, 'audio');
-            });
-        }
-
-        if (this.videoRecorder) {
-            this.videoRecorder.stopRecording(function (blobURL) {
-                callback(blobURL, 'video');
-            });
-        }
-
-        if (this.gifRecorder) {
-            this.gifRecorder.stopRecording(function (blobURL) {
-                callback(blobURL, 'gif');
-            });
-        }
-    };
-
-    this.getBlob = function (callback) {
-        var output = {};
-
-        if (this.audioRecorder) {
-            output.audio = this.audioRecorder.getBlob();
-        }
-
-        if (this.videoRecorder) {
-            output.video = this.videoRecorder.getBlob();
-        }
-
-        if (this.gifRecorder) {
-            output.gif = this.gifRecorder.getBlob();
-        }
-        if (callback) callback(output);
-    };
-
-    this.getDataURL = function (callback) {
-        this.getBlob(function (blob) {
-            getDataURL(blob.audio, function (_audioDataURL) {
-                getDataURL(blob.video, function (_videoDataURL) {
-                    callback({
-                        audio: _audioDataURL,
-                        video: _videoDataURL
-                    });
-                });
-            });
-        });
-
-        function getDataURL(blob, callback00) {
-            if (!!window.Worker) {
-                var webWorker = processInWebWorker(function readFile(_blob) {
-                    postMessage(new FileReaderSync().readAsDataURL(_blob));
-                });
-
-                webWorker.onmessage = function (event) {
-                    callback00(event.data);
-                };
-
-                webWorker.postMessage(blob);
-            }
-            else {
-                var reader = new FileReader();
-                reader.readAsDataURL(blob);
-                reader.onload = function (event) {
-                    callback00(event.target.result);
-                };
-            }
-        }
-
-        function processInWebWorker(_function) {
-            var blob = URL.createObjectURL(new Blob([_function.toString(),
-                'this.onmessage =  function (e) {readFile(e.data);}'
-            ], {
-                type: 'application/javascript'
-            }));
-
-            var worker = new Worker(blob);
-            URL.revokeObjectURL(blob);
-            return worker;
-        }
-    };
-
-    this.writeToDisk = function () {
-        RecordRTC.writeToDisk({
-            audio: this.audioRecorder,
-            video: this.videoRecorder,
-            gif: this.gifRecorder
-        });
-    };
-}
-
-MRecordRTC.getFromDisk = RecordRTC.getFromDisk;
-MRecordRTC.writeToDisk = RecordRTC.writeToDisk;
-
-// _____________________________
-// Cross-Browser-Declarations.js
-
-// animation-frame used in WebM recording
-if (!window.requestAnimationFrame) {
-    requestAnimationFrame = window.webkitRequestAnimationFrame || window.mozRequestAnimationFrame;
-}
-
-if (!window.cancelAnimationFrame) {
-    cancelAnimationFrame = window.webkitCancelAnimationFrame || window.mozCancelAnimationFrame;
-}
-
-// WebAudio API representer
-if (!window.AudioContext) {
-    window.AudioContext = window.webkitAudioContext || window.mozAudioContext;
-}
-
-URL = window.URL || window.webkitURL;
-navigator.getUserMedia = navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
-
-if (window.webkitMediaStream) window.MediaStream = window.webkitMediaStream;
-
-IsChrome = !!navigator.webkitGetUserMedia;
-
-// Merge all other data-types except "function"
-
-function mergeProps(mergein, mergeto) {
-    mergeto = reformatProps(mergeto);
-    for (var t in mergeto) {
-        if (typeof mergeto[t] !== 'function') {
-            mergein[t] = mergeto[t];
-        }
-    }
-    return mergein;
-}
-
-function reformatProps(obj) {
-    var output = {};
-    for (var o in obj) {
-        if (o.indexOf('-') != -1) {
-            var splitted = o.split('-');
-            var name = splitted[0] + splitted[1].split('')[0].toUpperCase() + splitted[1].substr(1);
-            output[name] = obj[o];
-        } else output[o] = obj[o];
-    }
-    return output;
-}
-
-// __________ (used to handle stuff like http://goo.gl/xmE5eg) issue #129
-// Storage.js
-var Storage = {
-    AudioContext: window.AudioContext || window.webkitAudioContext
-};
-
-// ______________________
-// MediaStreamRecorder.js
-
-// todo: need to show alert boxes for incompatible cases
-// encoder only supports 48k/16k mono audio channel
-
-/**
- * Implementation of https://dvcs.w3.org/hg/dap/raw-file/default/media-stream-capture/MediaRecorder.html
- * The MediaRecorder accepts a mediaStream as input source passed from UA. When recorder starts,
- * a MediaEncoder will be created and accept the mediaStream as input source.
- * Encoder will get the raw data by track data changes, encode it by selected MIME Type, then store the encoded in EncodedBufferCache object.
- * The encoded data will be extracted on every timeslice passed from Start function call or by RequestData function.
- * Thread model:
- * When the recorder starts, it creates a "Media Encoder" thread to read data from MediaEncoder object and store buffer in EncodedBufferCache object.
- * Also extract the encoded data and create blobs on every timeslice passed from start function or RequestData function called by UA.
- */
-
-function MediaStreamRecorder(mediaStream) {
-    var self = this;
-
-    // if user chosen only audio option; and he tried to pass MediaStream with
-    // both audio and video tracks;
-    // using a dirty workaround to generate audio-only stream so that we can get audio/ogg output.
-    if (this.type == 'audio' && mediaStream.getVideoTracks && mediaStream.getVideoTracks().length) {
-        var context = new AudioContext();
-        var mediaStreamSource = context.createMediaStreamSource(mediaStream);
-
-        var destination = context.createMediaStreamDestination();
-        mediaStreamSource.connect(destination);
-
-        mediaStream = destination.stream;
-    }
-
-    var dataAvailable = false;
-    this.record = function () {
-        // http://dxr.mozilla.org/mozilla-central/source/content/media/MediaRecorder.cpp
-        // https://wiki.mozilla.org/Gecko:MediaRecorder
-        // https://dvcs.w3.org/hg/dap/raw-file/default/media-stream-capture/MediaRecorder.html
-
-        // starting a recording session; which will initiate "Reading Thread"
-        // "Reading Thread" are used to prevent main-thread blocking scenarios
-        mediaRecorder = new MediaRecorder(mediaStream);
-
-        // Dispatching OnDataAvailable Handler
-        mediaRecorder.ondataavailable = function (e) {
-            if (dataAvailable) return;
-
-            if (!e.data.size) {
-                console.warn('Recording of', e.data.type, 'failed.');
-                return;
-            }
-
-            // todo: need to check who commented following two lines and why?
-            // pull #118
-            // if (self.recordedBlob) self.recordedBlob = new Blob([self.recordedBlob, e.data], { type: e.data.type || 'audio/ogg' });
-
-            dataAvailable = true;
-            self.recordedBlob = new Blob([e.data], {
-                type: e.data.type || 'audio/ogg'
-            });
-            self.callback();
-        };
-
-        mediaRecorder.onerror = function (error) {
-            console.warn(error);
-            mediaRecorder.stop();
-            self.record(0);
-        };
-
-        // void start(optional long mTimeSlice)
-        // The interval of passing encoded data from EncodedBufferCache to onDataAvailable
-        // handler. "mTimeSlice < 0" means Session object does not push encoded data to
-        // onDataAvailable, instead, it passive wait the client side pull encoded data
-        // by calling requestData API.
-        mediaRecorder.start(0);
-
-        // Start recording. If timeSlice has been provided, mediaRecorder will
-        // raise a dataavailable event containing the Blob of collected data on every timeSlice milliseconds.
-        // If timeSlice isn't provided, UA should call the RequestData to obtain the Blob data, also set the mTimeSlice to zero.
-
-        if (self.onAudioProcessStarted) self.onAudioProcessStarted();
-    };
-
-    this.stop = function (callback) {
-        this.callback = callback;
-        // mediaRecorder.state == 'recording' means that media recorder is associated with "session"
-        // mediaRecorder.state == 'stopped' means that media recorder is detached from the "session" ... in this case; "session" will also be deleted.
-
-        if (mediaRecorder.state == 'recording') {
-            mediaRecorder.requestData();
-            mediaRecorder.stop();
-        }
-    };
-
-    // Reference to "MediaRecorder" object
-    var mediaRecorder;
-}
-
-// _________________
-// StereoRecorder.js
-
-function StereoRecorder(mediaStream) {
-    this.record = function () {
-        mediaRecorder = new StereoAudioRecorder(mediaStream, this);
-
-        var self = this;
-        mediaRecorder.onAudioProcessStarted = function () {
-            if (self.onAudioProcessStarted) self.onAudioProcessStarted();
-        };
-
-        mediaRecorder.record();
-    };
-
-    this.stop = function (callback) {
-        var self = this;
-        if (mediaRecorder)
-            mediaRecorder.stop(function () {
-                self.recordedBlob = mediaRecorder.recordedBlob;
-                callback();
-            });
-    };
-
-    // Reference to "StereoAudioRecorder" object
-    var mediaRecorder;
-}
-
-// source code from: http://typedarray.org/wp-content/projects/WebAudioRecorder/script.js
-// https://github.com/mattdiamond/Recorderjs#license-mit
-// ______________________
-// StereoAudioRecorder.js
-
-// In Chrome, when the javascript node is out of scope, the onaudioprocess callback stops firing. 
-// This leads to audio being significantly shorter than the generated video.
-var __stereoAudioRecorderJavacriptNode;
-
-function StereoAudioRecorder(mediaStream, root) {
-    // variables
-    var leftchannel = [];
-    var rightchannel = [];
-    var recording = false;
-    var recordingLength = 0;
-
-    this.record = function () {
-        // reset the buffers for the new recording
-        leftchannel.length = rightchannel.length = 0;
-        recordingLength = 0;
-
-        recording = true;
-    };
-
-    this.stop = function (callback) {
-        setTimeout(onRecordingStopped, 1000);
-
-        function onRecordingStopped() {
-            // stop recording
-            recording = false;
-
-            // flat the left and right channels down
-            var leftBuffer = mergeBuffers(leftchannel, recordingLength);
-            var rightBuffer = mergeBuffers(rightchannel, recordingLength);
-
-            // interleave both channels together
-            var interleaved = interleave(leftBuffer, rightBuffer);
-
-            // create our wav file
-            var buffer = new ArrayBuffer(44 + interleaved.length * 2);
-            var view = new DataView(buffer);
-
-            // RIFF chunk descriptor
-            writeUTFBytes(view, 0, 'RIFF');
-            view.setUint32(4, 44 + interleaved.length * 2, true);
-            writeUTFBytes(view, 8, 'WAVE');
-
-            // FMT sub-chunk
-            writeUTFBytes(view, 12, 'fmt ');
-            view.setUint32(16, 16, true);
-            view.setUint16(20, 1, true);
-
-            // stereo (2 channels)
-            view.setUint16(22, 2, true);
-            view.setUint32(24, sampleRate, true);
-            view.setUint32(28, sampleRate * 4, true);
-            view.setUint16(32, 4, true);
-            view.setUint16(34, 16, true);
-
-            // data sub-chunk
-            writeUTFBytes(view, 36, 'data');
-            view.setUint32(40, interleaved.length * 2, true);
-
-            // write the PCM samples
-            var lng = interleaved.length;
-            var index = 44;
-            volume = 1;
-            for (var i = 0; i < lng; i++) {
-                view.setInt16(index, interleaved[i] * (0x7FFF * volume), true);
-                index += 2;
-            }
-
-            // final binary blob
-            self.recordedBlob = new Blob([view], {
-                type: 'audio/wav'
-            });
-
-            // recorded audio length
-            self.length = recordingLength;
-
-            callback();
-
-            isAudioProcessStarted = false;
-        }
-
-        var self = this;
-    };
-
-    function interleave(leftChannel, rightChannel) {
-        var length = leftChannel.length + rightChannel.length;
-        var result = new Float32Array(length);
-
-        var inputIndex = 0;
-
-        for (var index = 0; index < length;) {
-            result[index++] = leftChannel[inputIndex];
-            result[index++] = rightChannel[inputIndex];
-            inputIndex++;
-        }
-        return result;
-    }
-
-    function mergeBuffers(channelBuffer, rLength) {
-        var result = new Float32Array(rLength);
-        var offset = 0;
-        var lng = channelBuffer.length;
-        for (var i = 0; i < lng; i++) {
-            var buffer = channelBuffer[i];
-            result.set(buffer, offset);
-            offset += buffer.length;
-        }
-        return result;
-    }
-
-    function writeUTFBytes(view, offset, string) {
-        var lng = string.length;
-        for (var i = 0; i < lng; i++) {
-            view.setUint8(offset + i, string.charCodeAt(i));
-        }
-    }
-
-    // creates the audio context
-    var audioContext = Storage.AudioContext;
-
-    if (!Storage.AudioContextConstructor)
-        Storage.AudioContextConstructor = new audioContext();
-
-    var context = Storage.AudioContextConstructor;
-
-    // creates a gain node
-    if (!Storage.VolumeGainNode)
-        Storage.VolumeGainNode = context.createGain();
-
-    var volume = Storage.VolumeGainNode;
-
-    // creates an audio node from the microphone incoming stream
-    if (!Storage.AudioInput)
-        Storage.AudioInput = context.createMediaStreamSource(mediaStream);
-
-    var audioInput = Storage.AudioInput;
-
-    // connect the stream to the gain node
-    audioInput.connect(volume);
-
-    // From the spec: This value controls how frequently the audioprocess event is 
-    // dispatched and how many sample-frames need to be processed each call. 
-    // Lower values for buffer size will result in a lower (better) latency. 
-    // Higher values will be necessary to avoid audio breakup and glitches
-
-    // bug: how to minimize wav size?
-    // workaround? obviously ffmpeg!
-
-    // The size of the buffer (in sample-frames) which needs to 
-    // be processed each time onprocessaudio is called. 
-    // Legal values are (256, 512, 1024, 2048, 4096, 8192, 16384). 
-    var legalBufferValues = [256, 512, 1024, 2048, 4096, 8192, 16384];
-    var bufferSize = root.bufferSize || 4096;
-
-    if (legalBufferValues.indexOf(bufferSize) == -1) {
-        throw 'Legal values for buffer-size are ' + JSON.stringify(legalBufferValues, null, '\t');
-    }
-
-    // The sample rate (in sample-frames per second) at which the 
-    // AudioContext handles audio. It is assumed that all AudioNodes 
-    // in the context run at this rate. In making this assumption, 
-    // sample-rate converters or "varispeed" processors are not supported 
-    // in real-time processing.
-
-    // The sampleRate parameter describes the sample-rate of the 
-    // linear PCM audio data in the buffer in sample-frames per second. 
-    // An implementation must support sample-rates in at least 
-    // the range 22050 to 96000.
-    var sampleRate = root.sampleRate || context.sampleRate || 44100;
-
-    if (sampleRate < 22050 || sampleRate > 96000) {
-        throw 'sample-rate must be under range 22050 and 96000.';
-    }
-
-    console.log('sample-rate', sampleRate);
-    console.log('buffer-size', bufferSize);
-
-    if (context.createJavaScriptNode) {
-        __stereoAudioRecorderJavacriptNode = context.createJavaScriptNode(bufferSize, 2, 2);
-    } else if (context.createScriptProcessor) {
-        __stereoAudioRecorderJavacriptNode = context.createScriptProcessor(bufferSize, 2, 2);
-    } else {
-        throw 'WebAudio API has no support on this browser.';
-    }
-
-    var isAudioProcessStarted = false,
-        self = this;
-    __stereoAudioRecorderJavacriptNode.onaudioprocess = function (e) {
-        // if MediaStream().stop() or MediaStreamTrack().stop() is invoked.
-        if (mediaStream.ended) {
-            __stereoAudioRecorderJavacriptNode.onaudioprocess = function () {};
-            return;
-        }
-
-        if (!recording) return;
-
-        if (!isAudioProcessStarted) {
-            isAudioProcessStarted = true;
-            if (self.onAudioProcessStarted) {
-                self.onAudioProcessStarted();
-            }
-        }
-
-        var left = e.inputBuffer.getChannelData(0);
-        var right = e.inputBuffer.getChannelData(1);
-
-        // we clone the samples
-        leftchannel.push(new Float32Array(left));
-        rightchannel.push(new Float32Array(right));
-
-        recordingLength += bufferSize;
-    };
-
-    // we connect the recorder
-    volume.connect(__stereoAudioRecorderJavacriptNode);
-
-    // to prevent self audio to be connected with speakers
-    __stereoAudioRecorderJavacriptNode.connect(context.destination);
-}
-
-// _________________
-// CanvasRecorder.js
-
-function CanvasRecorder(htmlElement) {
-    if (!window.html2canvas) throw 'Please link: //www.webrtc-experiment.com/screenshot.js';
-
-    var isRecording;
-    this.record = function () {
-        isRecording = true;
-        drawCanvasFrame();
-    };
-
-    this.stop = function (callback) {
-        isRecording = false;
-        whammy.frames = dropFirstFrame(frames);
-
-        this.recordedBlob = whammy.compile();
-
-        frames = [];
-        if (callback) callback(this.recordedBlob);
-    };
-
-    var frames = [];
-
-    function drawCanvasFrame() {
-        html2canvas(htmlElement, {
-            onrendered: function (canvas) {
-                var duration = new Date().getTime() - lastTime;
-                if (!duration) return drawCanvasFrame();
-
-                // via #206, by Jack i.e. @Seymourr
-                lastTime = new Date().getTime();
-
-                frames.push({
-                    duration: duration,
-                    image: canvas.toDataURL('image/webp')
-                });
-
-                if (isRecording) requestAnimationFrame(drawCanvasFrame);
-            }
-        });
-    }
-
-    var lastTime = new Date().getTime();
-
-    var whammy = new Whammy.Video(100);
-}
-
-// _________________
-// WhammyRecorder.js
-
-function WhammyRecorder(mediaStream) {
-    this.record = function () {
-        if (!this.width) this.width = video.offsetWidth || 320;
-        if (!this.height) this.height = video.offsetHeight || 240;
-
-        if (!this.video) {
-            this.video = {
-                width: this.width,
-                height: this.height
-            };
-        }
-
-        if (!this.canvas) {
-            this.canvas = {
-                width: this.width,
-                height: this.height
-            };
-        }
-
-        canvas.width = this.canvas.width;
-        canvas.height = this.canvas.height;
-
-        video.width = this.video.width;
-        video.height = this.video.height;
-
-        console.log('canvas width', canvas.width);
-        console.log('canvas height', canvas.height);
-
-        console.log('video width', video.width);
-        console.log('video height', video.height);
-
-        context = canvas.getContext('2d');
-
-        drawFrames();
-    };
-
-    var frames = [];
-
-    // if user want to display advertisement before recorded video!
-    if (this.advertisement) {
-        frames = advertisement;
-    }
-
-    function drawFrames() {
-        var duration = new Date().getTime() - lastTime;
-        if (!duration) return drawFrames();
-
-        // via #206, by Jack i.e. @Seymourr
-        lastTime = new Date().getTime();
-
-        context.drawImage(video, 0, 0, canvas.width, canvas.height);
-        frames.push({
-            duration: duration,
-            image: canvas.toDataURL('image/webp')
-        });
-
-        if (!isStopDrawing) {
-            setTimeout(drawFrames, 10);
-        }
-    }
-
-    var isStopDrawing = false;
-
-    this.stop = function (callback) {
-        isStopDrawing = true;
-        whammy.frames = dropFirstFrame(frames);
-        frames = [];
-
-        this.recordedBlob = whammy.compile();
-
-        if (callback) callback(this.recordedBlob);
-    };
-
-    var canvas = document.createElement('canvas');
-    var context = canvas.getContext('2d');
-
-    var video = document.createElement('video');
-    video.muted = true;
-    video.volume = 0;
-    video.autoplay = true;
-    video.src = URL.createObjectURL(mediaStream);
-    video.play();
-
-    var lastTime = new Date().getTime();
-
-    var whammy = new Whammy.Video();
-}
-
-// https://github.com/antimatter15/whammy/blob/master/LICENSE
-// _________
-// Whammy.js
-
-// todo: Firefox now supports webp for webm containers!
-// their MediaRecorder implementation works well!
-// should we provide an option to record via Whammy.js or MediaRecorder API is a better solution?
-
-var Whammy = (function () {
-
-    function toWebM(frames) {
-        var info = checkFrames(frames);
-
-        var CLUSTER_MAX_DURATION = 30000;
-
-        var EBML = [{
-            "id": 0x1a45dfa3, // EBML
-            "data": [{
-                "data": 1,
-                "id": 0x4286 // EBMLVersion
-            }, {
-                "data": 1,
-                "id": 0x42f7 // EBMLReadVersion
-            }, {
-                "data": 4,
-                "id": 0x42f2 // EBMLMaxIDLength
-            }, {
-                "data": 8,
-                "id": 0x42f3 // EBMLMaxSizeLength
-            }, {
-                "data": "webm",
-                "id": 0x4282 // DocType
-            }, {
-                "data": 2,
-                "id": 0x4287 // DocTypeVersion
-            }, {
-                "data": 2,
-                "id": 0x4285 // DocTypeReadVersion
-            }]
-        }, {
-            "id": 0x18538067, // Segment
-            "data": [{
-                "id": 0x1549a966, // Info
-                "data": [{
-                    "data": 1e6, //do things in millisecs (num of nanosecs for duration scale)
-                    "id": 0x2ad7b1 // TimecodeScale
-                }, {
-                    "data": "whammy",
-                    "id": 0x4d80 // MuxingApp
-                }, {
-                    "data": "whammy",
-                    "id": 0x5741 // WritingApp
-                }, {
-                    "data": doubleToString(info.duration),
-                    "id": 0x4489 // Duration
-                }]
-            }, {
-                "id": 0x1654ae6b, // Tracks
-                "data": [{
-                    "id": 0xae, // TrackEntry
-                    "data": [{
-                        "data": 1,
-                        "id": 0xd7 // TrackNumber
-                    }, {
-                        "data": 1,
-                        "id": 0x63c5 // TrackUID
-                    }, {
-                        "data": 0,
-                        "id": 0x9c // FlagLacing
-                    }, {
-                        "data": "und",
-                        "id": 0x22b59c // Language
-                    }, {
-                        "data": "V_VP8",
-                        "id": 0x86 // CodecID
-                    }, {
-                        "data": "VP8",
-                        "id": 0x258688 // CodecName
-                    }, {
-                        "data": 1,
-                        "id": 0x83 // TrackType
-                    }, {
-                        "id": 0xe0, // Video
-                        "data": [{
-                            "data": info.width,
-                            "id": 0xb0 // PixelWidth
-                        }, {
-                            "data": info.height,
-                            "id": 0xba // PixelHeight
-                        }]
-                    }]
-                }]
-            }]
-        }];
-
-        //Generate clusters (max duration)
-        var frameNumber = 0;
-        var clusterTimecode = 0;
-        while (frameNumber < frames.length) {
-
-            var clusterFrames = [];
-            var clusterDuration = 0;
-            do {
-                clusterFrames.push(frames[frameNumber]);
-                clusterDuration += frames[frameNumber].duration;
-                frameNumber++;
-            } while (frameNumber < frames.length && clusterDuration < CLUSTER_MAX_DURATION);
-
-            var clusterCounter = 0;
-            var cluster = {
-                "id": 0x1f43b675, // Cluster
-                "data": [{
-                    "data": clusterTimecode,
-                    "id": 0xe7 // Timecode
-                }].concat(clusterFrames.map(function (webp) {
-                    var block = makeSimpleBlock({
-                        discardable: 0,
-                        frame: webp.data.slice(4),
-                        invisible: 0,
-                        keyframe: 1,
-                        lacing: 0,
-                        trackNum: 1,
-                        timecode: Math.round(clusterCounter)
-                    });
-                    clusterCounter += webp.duration;
-                    return {
-                        data: block,
-                        id: 0xa3
-                    };
-                }))
-            }; //Add cluster to segment
-            EBML[1].data.push(cluster);
-            clusterTimecode += clusterDuration;
-        }
-
-        return generateEBML(EBML);
-    }
-
-    // sums the lengths of all the frames and gets the duration
-
-    function checkFrames(frames) {
-        if (!frames[0]) {
-            console.warn('Something went wrong. Maybe WebP format is not supported in the current browser.');
-            return;
-        }
-
-        var width = frames[0].width,
-            height = frames[0].height,
-            duration = frames[0].duration;
-
-        for (var i = 1; i < frames.length; i++) {
-            duration += frames[i].duration;
-        }
-        return {
-            duration: duration,
-            width: width,
-            height: height
-        };
-    }
-
-    function numToBuffer(num) {
-        var parts = [];
-        while (num > 0) {
-            parts.push(num & 0xff);
-            num = num >> 8;
-        }
-        return new Uint8Array(parts.reverse());
-    }
-
-    function strToBuffer(str) {
-        return new Uint8Array(str.split('').map(function (e) {
-            return e.charCodeAt(0);
-        }));
-    }
-
-    function bitsToBuffer(bits) {
-        var data = [];
-        var pad = (bits.length % 8) ? (new Array(1 + 8 - (bits.length % 8))).join('0') : '';
-        bits = pad + bits;
-        for (var i = 0; i < bits.length; i += 8) {
-            data.push(parseInt(bits.substr(i, 8), 2));
-        }
-        return new Uint8Array(data);
-    }
-
-    function generateEBML(json) {
-        var ebml = [];
-        for (var i = 0; i < json.length; i++) {
-            var data = json[i].data;
-            if (typeof data == 'object') data = generateEBML(data);
-            if (typeof data == 'number') data = bitsToBuffer(data.toString(2));
-            if (typeof data == 'string') data = strToBuffer(data);
-
-            var len = data.size || data.byteLength || data.length;
-            var zeroes = Math.ceil(Math.ceil(Math.log(len) / Math.log(2)) / 8);
-            var size_str = len.toString(2);
-            var padded = (new Array((zeroes * 7 + 7 + 1) - size_str.length)).join('0') + size_str;
-            var size = (new Array(zeroes)).join('0') + '1' + padded;
-
-            ebml.push(numToBuffer(json[i].id));
-            ebml.push(bitsToBuffer(size));
-            ebml.push(data);
-        }
-
-        return new Blob(ebml, {
-            type: "video/webm"
-        });
-    }
-
-    function toBinStr_old(bits) {
-        var data = '';
-        var pad = (bits.length % 8) ? (new Array(1 + 8 - (bits.length % 8))).join('0') : '';
-        bits = pad + bits;
-        for (var i = 0; i < bits.length; i += 8) {
-            data += String.fromCharCode(parseInt(bits.substr(i, 8), 2));
-        }
-        return data;
-    }
-
-    function generateEBML_old(json) {
-        var ebml = '';
-        for (var i = 0; i < json.length; i++) {
-            var data = json[i].data;
-            if (typeof data == 'object') data = generateEBML_old(data);
-            if (typeof data == 'number') data = toBinStr_old(data.toString(2));
-
-            var len = data.length;
-            var zeroes = Math.ceil(Math.ceil(Math.log(len) / Math.log(2)) / 8);
-            var size_str = len.toString(2);
-            var padded = (new Array((zeroes * 7 + 7 + 1) - size_str.length)).join('0') + size_str;
-            var size = (new Array(zeroes)).join('0') + '1' + padded;
-
-            ebml += toBinStr_old(json[i].id.toString(2)) + toBinStr_old(size) + data;
-
-        }
-        return ebml;
-    }
-
-    function makeSimpleBlock(data) {
-        var flags = 0;
-        if (data.keyframe) flags |= 128;
-        if (data.invisible) flags |= 8;
-        if (data.lacing) flags |= (data.lacing << 1);
-        if (data.discardable) flags |= 1;
-        if (data.trackNum > 127) {
-            throw "TrackNumber > 127 not supported";
-        }
-        var out = [data.trackNum | 0x80, data.timecode >> 8, data.timecode & 0xff, flags].map(function (e) {
-            return String.fromCharCode(e);
-        }).join('') + data.frame;
-
-        return out;
-    }
-
-    function parseWebP(riff) {
-        var VP8 = riff.RIFF[0].WEBP[0];
-
-        var frame_start = VP8.indexOf('\x9d\x01\x2a'); // A VP8 keyframe starts with the 0x9d012a header
-        for (var i = 0, c = []; i < 4; i++) c[i] = VP8.charCodeAt(frame_start + 3 + i);
-
-        var width, height, tmp;
-
-        //the code below is literally copied verbatim from the bitstream spec
-        tmp = (c[1] << 8) | c[0];
-        width = tmp & 0x3FFF;
-        tmp = (c[3] << 8) | c[2];
-        height = tmp & 0x3FFF;
-        return {
-            width: width,
-            height: height,
-            data: VP8,
-            riff: riff
-        };
-    }
-
-    function parseRIFF(string) {
-        var offset = 0;
-        var chunks = {};
-
-        while (offset < string.length) {
-            var id = string.substr(offset, 4);
-            var len = parseInt(string.substr(offset + 4, 4).split('').map(function (i) {
-                var unpadded = i.charCodeAt(0).toString(2);
-                return (new Array(8 - unpadded.length + 1)).join('0') + unpadded;
-            }).join(''), 2);
-            var data = string.substr(offset + 4 + 4, len);
-            offset += 4 + 4 + len;
-            chunks[id] = chunks[id] || [];
-
-            if (id == 'RIFF' || id == 'LIST') {
-                chunks[id].push(parseRIFF(data));
-            } else {
-                chunks[id].push(data);
-            }
-        }
-        return chunks;
-    }
-
-    function doubleToString(num) {
-        return [].slice.call(
-            new Uint8Array((new Float64Array([num])).buffer), 0).map(function (e) {
-            return String.fromCharCode(e);
-        }).reverse().join('');
-    }
-
-    // a more abstract-ish API
-
-    function WhammyVideo(duration) {
-        this.frames = [];
-        this.duration = duration || 1;
-        this.quality = 100;
-    }
-
-    WhammyVideo.prototype.add = function (frame, duration) {
-        if ('canvas' in frame) { //CanvasRenderingContext2D
-            frame = frame.canvas;
-        }
-
-        if ('toDataURL' in frame) {
-            frame = frame.toDataURL('image/webp', this.quality);
-        }
-
-        if (!(/^data:image\/webp;base64,/ig).test(frame)) {
-            throw "Input must be formatted properly as a base64 encoded DataURI of type image/webp";
-        }
-        this.frames.push({
-            image: frame,
-            duration: duration || this.duration
-        });
-    };
-    WhammyVideo.prototype.compile = function () {
-        return new toWebM(this.frames.map(function (frame) {
-            var webp = parseWebP(parseRIFF(atob(frame.image.slice(23))));
-            webp.duration = frame.duration;
-            return webp;
-        }));
-    };
-    return {
-        Video: WhammyVideo,
-        toWebM: toWebM
-    };
-})();
-
-// ______________ (indexed-db)
-// DiskStorage.js
-
-var DiskStorage = {
-    init: function () {
-        var self = this;
-        var indexedDB = window.indexedDB || window.webkitIndexedDB || window.mozIndexedDB || window.OIndexedDB || window.msIndexedDB;
-        var dbVersion = 1;
-        var dbName = location.href.replace(/\/|:|#|%|\.|\[|\]/g, ''),
-            db;
-        var request = indexedDB.open(dbName, dbVersion);
-
-        function createObjectStore(dataBase) {
-            dataBase.createObjectStore(self.dataStoreName);
-        }
-
-        function putInDB() {
-            var transaction = db.transaction([self.dataStoreName], 'readwrite');
-
-            if (self.videoBlob) {
-                transaction.objectStore(self.dataStoreName).put(self.videoBlob, 'videoBlob');
-            }
-
-            if (self.gifBlob) {
-                transaction.objectStore(self.dataStoreName).put(self.gifBlob, 'gifBlob');
-            }
-
-            if (self.audioBlob) {
-                transaction.objectStore(self.dataStoreName).put(self.audioBlob, 'audioBlob');
-            }
-
-            function getFromStore(portionName) {
-                transaction.objectStore(self.dataStoreName).get(portionName).onsuccess = function (event) {
-                    if (self.callback) self.callback(event.target.result, portionName);
-                };
-            }
-
-            getFromStore('audioBlob');
-            getFromStore('videoBlob');
-            getFromStore('gifBlob');
-        }
-
-        request.onerror = self.onError;
-
-        request.onsuccess = function () {
-            db = request.result;
-            db.onerror = self.onError;
-
-            if (db.setVersion) {
-                if (db.version != dbVersion) {
-                    var setVersion = db.setVersion(dbVersion);
-                    setVersion.onsuccess = function () {
-                        createObjectStore(db);
-                        putInDB();
-                    };
-                } else {
-                    putInDB();
-                }
-            } else {
-                putInDB();
-            }
-        };
-        request.onupgradeneeded = function (event) {
-            createObjectStore(event.target.result);
-        };
-    },
-    Fetch: function (callback) {
-        this.callback = callback;
-        this.init();
-
-        return this;
-    },
-    Store: function (config) {
-        this.audioBlob = config.audioBlob;
-        this.videoBlob = config.videoBlob;
-        this.gifBlob = config.gifBlob;
-
-        this.init();
-
-        return this;
-    },
-    onError: function (error) {
-        console.error(JSON.stringify(error, null, '\t'));
-    },
-    dataStoreName: 'recordRTC'
-};
-
-// ______________
-// GifRecorder.js
-
-function GifRecorder(mediaStream) {
-    this.record = function () {
-        if (!this.width) this.width = video.offsetWidth || 320;
-        if (!this.height) this.height = video.offsetHeight || 240;
-
-        if (!this.video) {
-            this.video = {
-                width: this.width,
-                height: this.height
-            };
-        }
-
-        if (!this.canvas) {
-            this.canvas = {
-                width: this.width,
-                height: this.height
-            };
-        }
-
-        canvas.width = this.canvas.width;
-        canvas.height = this.canvas.height;
-
-        video.width = this.video.width;
-        video.height = this.video.height;
-
-        // external library to record as GIF images
-        gifEncoder = new GIFEncoder();
-
-        // void setRepeat(int iter) 
-        // Sets the number of times the set of GIF frames should be played. 
-        // Default is 1; 0 means play indefinitely.
-        gifEncoder.setRepeat(0);
-
-        // void setFrameRate(Number fps) 
-        // Sets frame rate in frames per second. 
-        // Equivalent to setDelay(1000/fps).
-        // Using "setDelay" instead of "setFrameRate"
-        gifEncoder.setDelay(this.frameRate || 200);
-
-        // void setQuality(int quality) 
-        // Sets quality of color quantization (conversion of images to the 
-        // maximum 256 colors allowed by the GIF specification). 
-        // Lower values (minimum = 1) produce better colors, 
-        // but slow processing significantly. 10 is the default, 
-        // and produces good color mapping at reasonable speeds. 
-        // Values greater than 20 do not yield significant improvements in speed.
-        gifEncoder.setQuality(this.quality || 10);
-
-        // Boolean start() 
-        // This writes the GIF Header and returns false if it fails.
-        gifEncoder.start();
-
-        startTime = Date.now();
-
-        function drawVideoFrame(time) {
-            lastAnimationFrame = requestAnimationFrame(drawVideoFrame);
-
-            if (typeof lastFrameTime === undefined) {
-                lastFrameTime = time;
-            }
-
-            // ~10 fps
-            if (time - lastFrameTime < 90) return;
-
-            context.drawImage(video, 0, 0, canvas.width, canvas.height);
-            gifEncoder.addFrame(context);
-            lastFrameTime = time;
-        }
-
-        lastAnimationFrame = requestAnimationFrame(drawVideoFrame);
-    };
-
-    this.stop = function () {
-        if (lastAnimationFrame) cancelAnimationFrame(lastAnimationFrame);
-
-        endTime = Date.now();
-
-        this.recordedBlob = new Blob([new Uint8Array(gifEncoder.stream().bin)], {
-            type: 'image/gif'
-        });
-
-        // bug: find a way to clear old recorded blobs
-        gifEncoder.stream().bin = [];
-    };
-
-    var canvas = document.createElement('canvas');
-    var context = canvas.getContext('2d');
-
-    var video = document.createElement('video');
-    video.muted = true;
-    video.autoplay = true;
-    video.src = URL.createObjectURL(mediaStream);
-    video.play();
-
-    var lastAnimationFrame = null;
-    var startTime, endTime, lastFrameTime;
-
-    var gifEncoder;
-}
-
-// This method is taken from a modified version of MediaStreamRecorder.js!
-// To solve first frame that is always blank. 
-// See: https://github.com/muaz-khan/WebRTC-Experiment/issues/94
-
-function dropFirstFrame(arr) {
-    arr.shift();
-    return arr;
-}
-
-if (location.href.indexOf('file:') == 0) {
-    console.error('Please load this HTML file on HTTP or HTTPS.');
-}
-
-// below function via: http://goo.gl/B3ae8c
-function bytesToSize(bytes) {
-    var k = 1000;
-    var sizes = ['Bytes', 'KB', 'MB', 'GB', 'TB'];
-    if (bytes === 0) return '0 Bytes';
-    var i = parseInt(Math.floor(Math.log(bytes) / Math.log(k)), 10);
-    return (bytes / Math.pow(k, i)).toPrecision(3) + ' ' + sizes[i];
-}
-
-// _____________
-// gifEncoder.js
-
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diff --git a/RecordRTC-Development/Synced-Audio-Video.html b/RecordRTC-Development/Synced-Audio-Video.html
deleted file mode 100644
index 2c8c835c..00000000
--- a/RecordRTC-Development/Synced-Audio-Video.html
+++ /dev/null
@@ -1,152 +0,0 @@
-<!DOCTYPE html>
-<html lang="en">
-    <head>
-        <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">
-        <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        
-        <style>
-            body, html {
-                background: black;
-                color: white;
-                text-align: center;
-            }
-
-            button, a {
-                background: -webkit-gradient(linear, 50% 0, 50% 100%, color-stop(0%, #fff), color-stop(100%, #eaeaea));
-                background: -webkit-linear-gradient(top, #fff, #eaeaea);
-                background: -moz-linear-gradient(top, #fff, #eaeaea);
-                background: -o-linear-gradient(top, #fff, #eaeaea);
-                background: linear-gradient(top, #fff, #eaeaea);
-                border: 1px solid white;
-                border-radius: 5px;
-                color: #4f4f4f;
-                padding: 8px 15px;
-                text-decoration: none;
-                text-transform: capitalize;
-            }
-
-            button:hover, a:hover, button:active, a:active, button:focus, a:focus {
-                -moz-box-shadow: inset 0 0 5px rgba(0, 0, 0, 0.5);
-                -webkit-box-shadow: inset 0 0 5px rgba(0, 0, 0, 0.5);
-                background: #303030;
-                border-color: white;
-                box-shadow: inset 0 0 5px rgba(0, 0, 0, 0.5);
-                color: white;
-            }
-
-            button[disabled] {
-                background: transparent;
-                border-color: rgb(83, 81, 81);
-                color: rgb(139, 133, 133);
-            }
-        </style>
-        <script src="RecordRTC.js"> </script>
-    </head>
-
-    <body style="text-align: center;">
-        <section class="experiment">  
-            <p style="text-align: center;">
-                <video id="preview" controls style="border: 1px solid rgb(15, 158, 238); height: 240px; max-width: 100%; vertical-align: top; width: 320px;"></video> 
-            </p>
-            
-            <p style="text-align: center;">
-                <audio controls style="border: 1px solid rgb(15, 158, 238); max-width: 100%; vertical-align: top; width: 320px;"></audio> 
-            </p>
-
-            <button id="record">Record</button>
-            <button id="stop" disabled>Stop</button>
-
-            <div id="container" style="padding: 1em 2em;"></div>
-        </section>
-            
-        <script>
-            function RenderBlobs(audioURL, videoURL) {
-                document.querySelector('audio').src = audioURL;
-                document.querySelector('video').src = videoURL;
-                
-                document.querySelector('audio').play();
-                document.querySelector('video').play();
-                
-                function mismatchChecker() {
-                   var mismatch = document.querySelector('audio').duration - document.querySelector('video').duration;
-                   stop.innerHTML = 'Audio has ' + (+mismatch.toFixed(1) * 1000) + ' milliseconds latency!';
-                };
-                
-                setTimeout(mismatchChecker, 2000);
-            }
-
-            var record = document.getElementById('record');
-            var stop = document.getElementById('stop');
-
-            var recordVideo = document.getElementById('record-video');
-            var preview = document.getElementById('preview');
-
-            var container = document.getElementById('container');
-
-            var isFirefox = !!navigator.mozGetUserMedia;
-
-            var recordAudio, recordVideo;
-            record.onclick = function() {
-                record.disabled = true;
-                !window.stream && navigator.getUserMedia({
-                        audio: true,
-                        video: true
-                    }, function(stream) {
-                        window.stream = stream;
-                        onstream();
-                    }, function(error) {
-                        alert(JSON.stringify(error, null, '\t'));
-                    });
-
-                window.stream && onstream();
-
-                function onstream() {
-                    preview.src = window.URL.createObjectURL(stream);
-                    preview.play();
-                    preview.muted = true;
-
-                    recordAudio = RecordRTC(stream, {
-                        bufferSize: 16384
-                    });
-                    recordAudio.startRecording();
-
-                    recordVideo = RecordRTC(stream, {
-                        type: 'video'
-                    });
-                    recordVideo.startRecording();
-
-                    stop.disabled = false;
-                }
-            };
-
-            var fileName;
-            stop.onclick = function() {
-                record.disabled = false;
-                stop.disabled = true;
-
-                preview.src = '';
-                preview.poster = '//www.webrtc-experiment.com/images/ajax-loader.gif';
-
-                fileName = Math.round(Math.random() * 99999999) + 99999999;
-
-                recordVideo.stopRecording(function(videoURL) {
-                    recordAudio.stopRecording(function(audioURL) {
-                       RenderBlobs(audioURL, videoURL);
-                    });
-                });
-            };
-        </script>
-        <br /><br /><hr style="height:3px; border-color:gray;"/>
-        <div style="text-align:left;">
-            <h2>Purpose of this experiment?</h2>
-            To make sure:
-            <ol>
-                <li>Length of both audio and video is same</li>
-                <li>There is no latency between audio and video playbacks!</li>
-                <li>In Chrome, when the javascript node is out of scope, the onaudioprocess callback stops firing. This leads to audio being significantly shorter than the generated video.</li>
-            </ol>
-        </div>
-    </body>
-</html>
diff --git a/RecordRTC-Development/audio-recording.html b/RecordRTC-Development/audio-recording.html
deleted file mode 100644
index 3594b2f6..00000000
--- a/RecordRTC-Development/audio-recording.html
+++ /dev/null
@@ -1,41 +0,0 @@
-<h1>
-    Record Audio using RecordRTC!
-    <button id="record">Record</button>
-    <button id="stop">Stop</button>
-</h1>
-
-<audio controls autoplay></audio>
-
-<script src="RecordRTC.js"> </script>
-<script>
-var mediaConstraints = { audio: true };
-
-var recordRTC;
-document.querySelector('#record').onclick = function() {
-    this.disabled = true;
-
-    navigator.getUserMedia(mediaConstraints, function(mediaStream) {
-        document.querySelector('audio').src = URL.createObjectURL(mediaStream);
-
-        recordRTC = RecordRTC(mediaStream, {
-            type: 'audio',
-            bufferSize: 4096,
-            sampleRate: 44100
-        });
-        recordRTC.startRecording();
-    }, onMediaError);
-};
-
-document.querySelector('#stop').onclick = function() {
-    this.disabled = true;
-
-    recordRTC.stopRecording(function() {
-        var blob = recordRTC.getBlob();
-        document.querySelector('audio').src = URL.createObjectURL(blob);
-    });
-};
-
-function onMediaError(error) {
-    alert(JSON.stringify(error, null, '\t'));
-}
-</script>
\ No newline at end of file
diff --git a/RecordRTC-Development/video-recording.html b/RecordRTC-Development/video-recording.html
deleted file mode 100644
index 8adcd329..00000000
--- a/RecordRTC-Development/video-recording.html
+++ /dev/null
@@ -1,39 +0,0 @@
-<h1>
-    Record Video using RecordRTC!
-    <button id="record">Record</button>
-    <button id="stop">Stop</button>
-</h1>
-
-<video controls autoplay width=320 height=240></video>
-
-<script src="RecordRTC.js"> </script>
-<script>
-var mediaConstraints = { video: true };
-
-var recordRTC;
-document.querySelector('#record').onclick = function() {
-    this.disabled = true;
-
-    navigator.getUserMedia(mediaConstraints, function(mediaStream) {
-        document.querySelector('video').src = URL.createObjectURL(mediaStream);
-
-        recordRTC = RecordRTC(mediaStream, {
-            type: 'video'
-        });
-        recordRTC.startRecording();
-    }, onMediaError);
-};
-
-document.querySelector('#stop').onclick = function() {
-    this.disabled = true;
-
-    recordRTC.stopRecording(function() {
-        var blob = recordRTC.getBlob();
-        document.querySelector('video').src = URL.createObjectURL(blob);
-    });
-};
-
-function onMediaError(error) {
-    alert(JSON.stringify(error, null, '\t'));
-}
-</script>
\ No newline at end of file
diff --git a/RecordRTC/AudioVideo-on-Firefox.html b/RecordRTC/AudioVideo-on-Firefox.html
index b62a4e5c..7bbed071 100644
--- a/RecordRTC/AudioVideo-on-Firefox.html
+++ b/RecordRTC/AudioVideo-on-Firefox.html
@@ -16,7 +16,7 @@
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
         
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             video {
@@ -51,7 +51,7 @@
         </script>
         
         <!-- script used for audio/video/gif recording -->
-        <script src="https://www.webrtc-experiment.com/RecordRTC.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RecordRTC.js"> </script>
     </head>
 
     <body>
@@ -208,6 +208,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/RecordRTC/README.md b/RecordRTC/README.md
index 2eb6ba96..0c61c170 100644
--- a/RecordRTC/README.md
+++ b/RecordRTC/README.md
@@ -55,7 +55,7 @@ There are some other NPM packages regarding RecordRTC:
 =
 
 ```html
-<script src="//www.WebRTC-Experiment.com/RecordRTC.js"></script>
+<script src="//cdn.WebRTC-Experiment.com/RecordRTC.js"></script>
 ```
 
 =
diff --git a/RecordRTC/index.html b/RecordRTC/index.html
index 434fd8c6..3aa1cf2d 100644
--- a/RecordRTC/index.html
+++ b/RecordRTC/index.html
@@ -13,7 +13,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio {
@@ -53,7 +53,7 @@
         </script>
         
         <!-- script used for audio/video/gif recording -->
-        <script src="https://www.webrtc-experiment.com/RecordRTC.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RecordRTC.js"> </script>
     </head>
 
     <body>
@@ -335,15 +335,6 @@ <h2 class="header">Try <a href="https://www.webrtc-experiment.com/RecordRTC/Audi
                     videoHeight_input.value = canvasHeight_input.value = innerHeight;
                 };
             </script>
-        
-            <section class="experiment">
-                <h2 class="header" id="feedback">Feedback</h2>
-                <div>
-                    <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><small style="margin-left: 1em;">Enter your email too; if you want "direct" reply!</small>
-            </section>
-			
 			
             <section class="experiment">
                 <h2 class="header">Using RecordRTC...</h2>
@@ -393,7 +384,7 @@ <h2 class="header">Technical Guide</h2>
                 <h2 class="header">
                     How to use <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC" target="_blank">RecordRTC</a>?</h2>
                 <pre>
-&lt;script src="https://www.WebRTC-Experiment.com/RecordRTC.js"&gt;&lt;/script&gt;
+&lt;script src="//cdn.WebRTC-Experiment.com/RecordRTC.js"&gt;&lt;/script&gt;
 </pre>
             </section>
             <section class="experiment">
@@ -484,6 +475,14 @@ <h2>
                     RecordRTC is MIT licensed on Github! <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC" target="_blank">Documentation</a>
                 </p>
             </section>
+            
+            <section class="experiment">
+                <h2 class="header" id="feedback">Feedback</h2>
+                <div>
+                    <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><small style="margin-left: 1em;">Enter your email too; if you want "direct" reply!</small>
+            </section>
 			
             <section class="experiment own-widgets latest-commits">
                 <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a href="https://github.com/muaz-khan/WebRTC-Experiment/commits/master" target="_blank">Latest Updates</a></h2>
@@ -502,6 +501,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/Translator.js/README.md b/Translator.js/README.md
index 561e9a62..9f98394c 100644
--- a/Translator.js/README.md
+++ b/Translator.js/README.md
@@ -7,7 +7,7 @@ Translator.js is a JavaScript library built top on Google Speech-Recognition & T
 ##### How to use?
 
 ```html
-<script src="//www.webrtc-experiment.com/Translator.js"> </script>
+<script src="//cdn.webrtc-experiment.com/Translator.js"> </script>
 ```
 
 =
@@ -72,7 +72,7 @@ This method uses meSpeak.js library to play text using a robot voice. Behind the
 
 ```javascript
 translator.speakTextUsingRobot(textToPlay, {
-    workerPath: 'https://www.webrtc-experiment.com/Robot-Speaker.js',
+    workerPath: '//cdn.webrtc-experiment.com/Robot-Speaker.js',
     callback: function (WAV_File) {},
     amplitude: 100,
     wordgap: 0,
diff --git a/Translator.js/Translator.js b/Translator.js/Translator.js
index c1acbb7c..2b5bdb95 100644
--- a/Translator.js/Translator.js
+++ b/Translator.js/Translator.js
@@ -1,4 +1,4 @@
-// Last time updated at 04 Feb 2014, 05:46:23
+// Last time updated at July 17, 2014, 05:46:23
 
 // Muaz Khan      - www.MuazKhan.com
 // MIT License    - www.WebRTC-Experiment.com/licence
@@ -60,6 +60,9 @@ function Translator() {
             if (response.data && response.data.translations[0] && config.callback) {
                 config.callback(response.data.translations[0].translatedText);
             }
+            if(response.error && response.error.message == 'Daily Limit Exceeded') {
+                config.callback('Google says, "Daily Limit Exceeded". Please try this experiment a few hours later.');
+            }
         };
 
         var source = 'https://www.googleapis.com/language/translate/v2?key=' + api_key + '&target=' + (config.to || 'en-US') + '&callback=window.' + randomNumber + '&q=' + sourceText;
diff --git a/Translator.js/index.html b/Translator.js/index.html
index de37c08e..e864ec07 100644
--- a/Translator.js/index.html
+++ b/Translator.js/index.html
@@ -10,17 +10,25 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="//www.webrtc-experiment.com/style.css">
+        
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
+        
         <script>
             document.createElement('article');
             document.createElement('footer');
         </script>
     
-        <script src="//www.webrtc-experiment.com/firebase.js"> </script>
-        <script src="//www.webrtc-experiment.com/Translator.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/Translator.js"> </script>
+        
+        <!-- scripts used for styling code -->
+        <script src="//cdn.webrtc-experiment.com/syntax/sh_main.min.js" type="text/javascript"> </script>
+        <script src="//cdn.webrtc-experiment.com/syntax/sh_javascript.min.js" type="text/javascript"> </script>
+        <script src="//cdn.webrtc-experiment.com/syntax/sh_html.min.js" type="text/javascript"> </script>
+        <link href="//cdn.webrtc-experiment.com/syntax/sh_style.css" type="text/css" rel="stylesheet">
     </head>
 
-    <body>
+    <body onload="sh_highlightDocument();">
         <article>
     
             <header style="text-align: center;">
@@ -58,7 +66,7 @@ <h2 class="header" id="what">What is Translator.js?</h2>
                 <h2 class="header" id="test">Test among two computers or windows:</h2>
                 <p>
                     <input type="checkbox" id="robot-voice" checked>
-                    <label for="robot-voice"><a href="https://www.webrtc-experiment.com/Robot-Speaker.js">Robot</a> Voice?</label>
+                    <label for="robot-voice"><a href="//cdn.webrtc-experiment.com/Robot-Speaker.js">Robot</a> Voice?</label>
 
                     <select id="your-language">
                         <option>en (English)</option>
@@ -147,6 +155,7 @@ <h2 class="header" id="how">How to test?</h2>
 
                     log('original language: ' + data.language);
                     log('original text: ' + data.text);
+                    log('converting into: ' + yourLanguage.value.split(' (')[0]);
 
                     translator.translateLanguage(data.text, {
                         from: data.language,
@@ -155,9 +164,10 @@ <h2 class="header" id="how">How to test?</h2>
                             log('translated text: ' + translatedText);
 
                             if (robotVoice.checked) {
+                                log('Using robot to speak text.');
                                 translator.speakTextUsingRobot(translatedText, {
                                     // please download and link worker file on your own domain
-                                    workerPath: 'https://www.webrtc-experiment.com/Robot-Speaker.js',
+                                    workerPath: 'https://cdn.webrtc-experiment.com/Robot-Speaker.js',
 									onWorkerFileDownloadStart: function() {
 										log('Downloading worker file. It is about 2MB in size.');
 									},
@@ -166,6 +176,7 @@ <h2 class="header" id="how">How to test?</h2>
 									}
                                 });
                             } else {
+                                log('Using Google Translate API to speak text.');
                                 translator.speakTextUsingGoogleSpeaker({
                                     textToSpeak: translatedText,
                                     targetLanguage: yourLanguage.value.split(' (')[0],
@@ -196,6 +207,79 @@ <h2 class="header" id="how">How to test?</h2>
 
             <br />
             <br />
+            
+            <section class="experiment">
+                <h2 class="header">
+                    How to use <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Translator.js" target="_blank">Translator.js</a>?</h2>
+                <pre class="sh_html">
+&lt;script src="//cdn.WebRTC-Experiment.com/Translator.js"&gt;&lt;/script&gt;
+</pre>
+            </section>
+            <section class="experiment">
+                <pre class="sh_javascript">
+// constructor
+var translator = new Translator();
+
+// voiceToText method allows you convert voice to text; 
+// whatever you speak is recognized using Google 
+// Speech-Recognition API; and converted into 
+// text using same API:
+translator.voiceToText(function (text) {
+    console.log('Your voice as text!', text);
+}, 'en-US');
+
+// translateLanguage method allows you convert language. 
+// You just need to pass text and get translated text 
+// through callback parameter:
+translator.translateLanguage(textToConvert, {
+    from: 'language-of-the-text',
+    to: 'convert-into',
+    callback: function (translatedText) {
+        console.log('translated text', translatedText);
+    }
+});
+
+// speakTextUsingRobot method uses meSpeak.js library 
+// to play text using a robot voice. Behind the scene; 
+// text is buffered and converted into WAV file; 
+// which is played using invisible &lt;audio&gt; element.
+translator.speakTextUsingRobot(textToPlay, {
+    workerPath: '//cdn.webrtc-experiment.com/Robot-Speaker.js',
+    callback: function (WAV_File) {},
+    amplitude: 100,
+    wordgap: 0,
+    pitch: 50,
+    speed: 175,
+    onSpeakingEnd: function() {},
+    onWorkerFileDownloadStart: function() {},
+    onWorkerFileDownloadEnd: function() {}
+});
+
+// If you want to play WAV file yourself or you want to 
+// POST/store WAV file; then you can use callback 
+// parameter to override default behaviour:
+translator.speakTextUsingRobot(textToPlay, {
+    callback: function (WAV_File) {
+        HTTP_POST_using_FormData( WAV_File );
+    }
+});
+
+// speakTextUsingGoogleSpeaker method uses Google's Non-Official 
+// Translation API to convert text into mp3 sound. API Key 
+// used is taken from someone on the web; and there is 
+// no guarantee of its availability; that's why it is 
+// strongly suggested to buy your own KEY and pass using 
+// api_key parameter:
+translator.speakTextUsingGoogleSpeaker({
+    textToSpeak: 'text-to-convert',
+    targetLanguage: 'your-language',
+
+    // Google Translation service's API Key
+    api_key: 'Your-Private-API-Key'
+});
+</pre>                
+            </section>
+            
             <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;">Want to ask a Question?</h2>
                 <div>
@@ -222,7 +306,7 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="//www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 
 </html>
diff --git a/ffmpeg/README.md b/ffmpeg/README.md
index 6e546e16..a999a8ff 100644
--- a/ffmpeg/README.md
+++ b/ffmpeg/README.md
@@ -1,8 +1,8 @@
 ### [ffmpeg-asm.js and RecordRTC! Audio/Video Merging & Transcoding!](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/ffmpeg) / [Demos](https://www.webrtc-experiment.com/ffmpeg/)
 
-[RecordRTC](https://www.webrtc-experiment.com/RecordRTC/) is used to record WAV/WebM files; [`ffmpeg-asm.js`](https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js) is used to merge/transcode in mp4 or ogg/mp3.
+[RecordRTC](https://www.webrtc-experiment.com/RecordRTC/) is used to record WAV/WebM files; [`ffmpeg-asm.js`](https://cdn.webrtc-experiment.com/ffmpeg_asm.js) is used to merge/transcode in mp4 or ogg/mp3.
 
-Remember: [`ffmpeg-asm.js`](https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js)'s credit goes to: https://github.com/bgrins/videoconverter.js
+Remember: [`ffmpeg-asm.js`](https://cdn.webrtc-experiment.com/ffmpeg_asm.js)'s credit goes to: https://github.com/bgrins/videoconverter.js
 
 =
 
@@ -13,7 +13,7 @@ Remember: [`ffmpeg-asm.js`](https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2
 3. [Transcoding WebM into mp4; then merging WAV+mp4 into single mp4](https://www.webrtc-experiment.com/ffmpeg/merging-wav-and-webm-into-mp4.html) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/ffmpeg/merging-wav-and-webm-into-mp4.html)
 4. [Recording Audio+Canvas and merging in single mp4](https://www.webrtc-experiment.com/ffmpeg/audio-plus-canvas-recording.html) / [Source Code](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/ffmpeg/audio-plus-canvas-recording.html)
 
-<img src="https://www.webrtc-experiment.com/images/audo-plus-canvas-recording.gif" />
+<img src="//cdn.webrtc-experiment.com/images/audo-plus-canvas-recording.gif" />
 
 =
 
@@ -21,7 +21,7 @@ Remember: [`ffmpeg-asm.js`](https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2
 
 **It is suggested to download `ffmpeg-asm` file and try locally!**
 
-[`ffmpeg-asm.js`](https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js) file's default size is 18MB; however you can gzip it to make it about 6MB.
+[`ffmpeg-asm.js`](https://cdn.webrtc-experiment.com/ffmpeg_asm.js) file's default size is 18MB; however you can gzip it to make it about 6MB.
 
 =
 
diff --git a/ffmpeg/audio-plus-canvas-recording.html b/ffmpeg/audio-plus-canvas-recording.html
index e1d34a86..086abc76 100644
--- a/ffmpeg/audio-plus-canvas-recording.html
+++ b/ffmpeg/audio-plus-canvas-recording.html
@@ -84,8 +84,8 @@ <h3>Content is edit-able.</h3>
     <button id="stop" disabled contenteditable="false">Stop</button>
 </div>
 
-<script src="https://www.webrtc-experiment.com/screenshot.js"> </script>
-<script src="https://www.webrtc-experiment.com/RecordRTC.js"> </script>
+<script src="//cdn.webrtc-experiment.com/screenshot.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RecordRTC.js"> </script>
 
 <script>
 var elementToShare = document.getElementById('elementToShare');
@@ -122,7 +122,7 @@ <h3>Content is edit-able.</h3>
 
 
 // var workerPath = location.href.replace(location.href.split('/').pop(), '') + 'ffmpeg_asm.js';
-var workerPath = 'https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js';
+var workerPath = 'https://cdn.webrtc-experiment.com/ffmpeg_asm.js';
 
 function processInWebWorker() {
     var blob = URL.createObjectURL(new Blob(['importScripts("' + workerPath + '");var now = Date.now;function print(text) {postMessage({"type" : "stdout","data" : text});};onmessage = function(event) {var message = event.data;if (message.type === "command") {var Module = {print: print,printErr: print,files: message.files || [],arguments: message.arguments || [],TOTAL_MEMORY: 268435456};postMessage({"type" : "start","data" : Module.arguments.join(" ")});postMessage({"type" : "stdout","data" : "Received command: " +Module.arguments.join(" ") +((Module.TOTAL_MEMORY) ? ".  Processing with " + Module.TOTAL_MEMORY + " bits." : "")});var time = now();var result = ffmpeg_run(Module);var totalTime = now() - time;postMessage({"type" : "stdout","data" : "Finished processing (took " + totalTime + "ms)"});postMessage({"type" : "done","data" : result,"time" : totalTime});}};postMessage({"type" : "ready"});'], {
@@ -198,7 +198,15 @@ <h3>Content is edit-able.</h3>
 
         worker.postMessage({
             type: 'command',
-            arguments: ['-i', 'video.webm', '-i', 'audio.wav', '-c:v', 'mpeg4', '-c:a', 'vorbis', '-b:v', '64k', '-b:a', '12k', '-strict', 'experimental', 'output.mp4'],
+            arguments: [
+                '-i', 'video.webm', 
+                '-i', 'audio.wav', 
+                '-c:v', 'mpeg4', 
+                '-c:a', 'vorbis', 
+                '-b:v', '6400k', 
+                '-b:a', '4800k', 
+                '-strict', 'experimental', 'output.mp4'
+            ],
             files: [
                 {
                     data: new Uint8Array(vab),
@@ -225,4 +233,6 @@ <h3>Content is edit-able.</h3>
     console.log(message);
 }
 </script>
-<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;">←WebRTC Experiments Homepage</a>
\ No newline at end of file
+<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;">←WebRTC Experiments Homepage</a>
+
+<script src="https://cdn.webrtc-experiment.com/ffmpeg_asm.js" async> </script>
\ No newline at end of file
diff --git a/ffmpeg/merging-wav-and-webm-into-mp4.html b/ffmpeg/merging-wav-and-webm-into-mp4.html
index 29a50bdb..2fad8c04 100644
--- a/ffmpeg/merging-wav-and-webm-into-mp4.html
+++ b/ffmpeg/merging-wav-and-webm-into-mp4.html
@@ -13,7 +13,8 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             video {
@@ -48,7 +49,7 @@
         </script>
         
         <!-- script used for audio/video/gif recording -->
-        <script src="//www.webrtc-experiment.com/RecordRTC.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RecordRTC.js"> </script>
     </head>
 
     <body>
@@ -137,13 +138,13 @@ <h2 class="header">Record audio/video and convert/merge into "mp4"!</h2>
                         recordVideo.stopRecording(function() {
                             convertStreams(recordVideo.getBlob(), recordAudio.getBlob());
 
-                            log('<a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file download started. It is about 18MB in size; please be patient!');
+                            log('<a href="'+ workerPath +'" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file download started. It is about 18MB in size; please be patient!');
                         });
                     });
                 };
 
                 // var workerPath = location.href.replace(location.href.split('/').pop(), '') + 'ffmpeg_asm.js';
-                var workerPath = 'https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js';
+                var workerPath = 'https://cdn.webrtc-experiment.com/ffmpeg_asm.js';
 
                 function processInWebWorker() {
                     var blob = URL.createObjectURL(new Blob(['importScripts("' + workerPath + '");var now = Date.now;function print(text) {postMessage({"type" : "stdout","data" : text});};onmessage = function(event) {var message = event.data;if (message.type === "command") {var Module = {print: print,printErr: print,files: message.files || [],arguments: message.arguments || [],TOTAL_MEMORY: message.TOTAL_MEMORY || false};postMessage({"type" : "start","data" : Module.arguments.join(" ")});postMessage({"type" : "stdout","data" : "Received command: " +Module.arguments.join(" ") +((Module.TOTAL_MEMORY) ? ".  Processing with " + Module.TOTAL_MEMORY + " bits." : "")});var time = now();var result = ffmpeg_run(Module);var totalTime = now() - time;postMessage({"type" : "stdout","data" : "Finished processing (took " + totalTime + "ms)"});postMessage({"type" : "done","data" : result,"time" : totalTime});}};postMessage({"type" : "ready"});'], {
@@ -191,14 +192,14 @@ <h2 class="header">Record audio/video and convert/merge into "mp4"!</h2>
                     worker.onmessage = function(event) {
                         var message = event.data;
                         if (message.type == "ready") {
-                            log('<a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file has been loaded.');
+                            log('<a href="'+ workerPath +'" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file has been loaded.');
                             workerReady = true;
                             if (buffersReady)
                                 postMessage();
                         } else if (message.type == "stdout") {
                             log(message.data);
                         } else if (message.type == "start") {
-                            log('<a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file received ffmpeg command.');
+                            log('<a href="'+ workerPath +'" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file received ffmpeg command.');
                         } else if (message.type == "done") {
                             log(JSON.stringify(message));
 
@@ -219,7 +220,15 @@ <h2 class="header">Record audio/video and convert/merge into "mp4"!</h2>
 
                         worker.postMessage({
                             type: 'command',
-                            arguments: ['-i', videoFile, '-i', 'audio.wav', '-c:v', 'mpeg4', '-c:a', 'vorbis', '-b:v', '64k', '-b:a', '12k', '-strict', 'experimental', 'output.mp4'],
+                            arguments: [
+                                '-i', videoFile, 
+                                '-i', 'audio.wav', 
+                                '-c:v', 'mpeg4', 
+                                '-c:a', 'vorbis', 
+                                '-b:v', '6400k', 
+                                '-b:a', '4800k', 
+                                '-strict', 'experimental', 'output.mp4'
+                            ],
                             files: [
                                 {
                                     data: new Uint8Array(vab),
@@ -309,6 +318,7 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script> 
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script> 
+        <script src="https://cdn.webrtc-experiment.com/ffmpeg_asm.js" async> </script>
     </body>
 </html>
diff --git a/ffmpeg/wav-to-ogg.html b/ffmpeg/wav-to-ogg.html
index 12f04ffb..ba3e8b05 100644
--- a/ffmpeg/wav-to-ogg.html
+++ b/ffmpeg/wav-to-ogg.html
@@ -13,7 +13,8 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio { vertical-align: middle; }
@@ -45,7 +46,7 @@
         </script>
         
         <!-- script used for audio/video/gif recording -->
-        <script src="//www.webrtc-experiment.com/RecordRTC.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RecordRTC.js"> </script>
     </head>
 
     <body>
@@ -85,7 +86,7 @@ <h2 class="header">Logs</h2>
             </section>
             
             <section class="experiment">  
-                <h2 class="header">Record Audio and convert into "Ogg" using <a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a>!</h2>
+                <h2 class="header">Record Audio and convert into "Ogg" using <a href="https://cdn.webrtc-experiment.com/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a>!</h2>
 			
                 <div class="inner">
                     <audio id="audio-preview" controls></audio>
@@ -122,13 +123,13 @@ <h2 class="header">Record Audio and convert into "Ogg" using <a href="https://go
                         audioPreview.src = url;
                         audioPreview.download = 'Orignal.wav';
                         
-                        log('<a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file download started. It is about 18MB in size; please be patient!');
+                        log('<a href="'+ workerPath +'" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file download started. It is about 18MB in size; please be patient!');
                         convertStreams(recordAudio.getBlob());
                     });
                 };
 
                 // var workerPath = location.href.replace(location.href.split('/').pop(), '') + 'ffmpeg_asm.js';
-                var workerPath = 'https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js';
+                var workerPath = 'https://cdn.webrtc-experiment.com/ffmpeg_asm.js';
 
                 function processInWebWorker() {
                     var blob = URL.createObjectURL(new Blob(['importScripts("' + workerPath + '");var now = Date.now;function print(text) {postMessage({"type" : "stdout","data" : text});};onmessage = function(event) {var message = event.data;if (message.type === "command") {var Module = {print: print,printErr: print,files: message.files || [],arguments: message.arguments || [],TOTAL_MEMORY: message.TOTAL_MEMORY || false};postMessage({"type" : "start","data" : Module.arguments.join(" ")});postMessage({"type" : "stdout","data" : "Received command: " +Module.arguments.join(" ") +((Module.TOTAL_MEMORY) ? ".  Processing with " + Module.TOTAL_MEMORY + " bits." : "")});var time = now();var result = ffmpeg_run(Module);var totalTime = now() - time;postMessage({"type" : "stdout","data" : "Finished processing (took " + totalTime + "ms)"});postMessage({"type" : "done","data" : result,"time" : totalTime});}};postMessage({"type" : "ready"});'], {
@@ -162,7 +163,7 @@ <h2 class="header">Record Audio and convert into "Ogg" using <a href="https://go
                     worker.onmessage = function(event) {
                         var message = event.data;
                         if (message.type == "ready") {
-                            log('<a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file has been loaded.');
+                            log('<a href="'+ workerPath +'" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file has been loaded.');
 
                             workerReady = true;
                             if (buffersReady)
@@ -170,7 +171,7 @@ <h2 class="header">Record Audio and convert into "Ogg" using <a href="https://go
                         } else if (message.type == "stdout") {
                             log(message.data);
                         } else if (message.type == "start") {
-                            log('<a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file received ffmpeg command.');
+                            log('<a href="'+ workerPath +'" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file received ffmpeg command.');
                         } else if (message.type == "done") {
                             log(JSON.stringify(message));
 
@@ -191,7 +192,12 @@ <h2 class="header">Record Audio and convert into "Ogg" using <a href="https://go
 
                         worker.postMessage({
                             type: 'command',
-                            arguments: ['-i', 'audio.wav', '-c:a', 'vorbis', '-b:a', '48k', '-strict', 'experimental', 'output.mp4'],
+                            arguments: [
+                                '-i', 'audio.wav', 
+                                '-c:a', 'vorbis', 
+                                '-b:a', '4800k', 
+                                '-strict', 'experimental', 'output.mp4'
+                            ],
                             files: [
                                 {
                                     data: new Uint8Array(aab),
@@ -288,6 +294,7 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
+        <script src="https://cdn.webrtc-experiment.com/ffmpeg_asm.js" async> </script>
     </body>
 </html>
diff --git a/ffmpeg/webm-to-mp4.html b/ffmpeg/webm-to-mp4.html
index df1ffba7..b6173b97 100644
--- a/ffmpeg/webm-to-mp4.html
+++ b/ffmpeg/webm-to-mp4.html
@@ -13,7 +13,8 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             video {
@@ -48,7 +49,7 @@
         </script>
         
         <!-- script used for audio/video/gif recording -->
-        <script src="//www.webrtc-experiment.com/RecordRTC.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RecordRTC.js"> </script>
     </head>
 
     <body>
@@ -88,7 +89,7 @@ <h2 class="header">Logs</h2>
             </section>
             
             <section class="experiment">  
-                <h2 class="header">Record Video and convert into "mp4" using <a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a>!</h2>
+                <h2 class="header">Record Video and convert into "mp4" using <a href="https://cdn.webrtc-experiment.com/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a>!</h2>
 			
                 <div class="inner">
                     <video id="video-preview" controls></video>
@@ -126,13 +127,13 @@ <h2 class="header">Record Video and convert into "mp4" using <a href="https://go
                         videoPreview.src = url;
                         videoPreview.download = videoFile;
                         
-                        log('<a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file download started. It is about 18MB in size; please be patient!');
+                        log('<a href="'+ workerPath +'" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file download started. It is about 18MB in size; please be patient!');
                         convertStreams(recordVideo.getBlob());
                     });
                 };
 
                 // var workerPath = location.href.replace(location.href.split('/').pop(), '') + 'ffmpeg_asm.js';
-                var workerPath = 'https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js';
+                var workerPath = 'https://cdn.webrtc-experiment.com/ffmpeg_asm.js';
 
                 function processInWebWorker() {
                     var blob = URL.createObjectURL(new Blob(['importScripts("' + workerPath + '");var now = Date.now;function print(text) {postMessage({"type" : "stdout","data" : text});};onmessage = function(event) {var message = event.data;if (message.type === "command") {var Module = {print: print,printErr: print,files: message.files || [],arguments: message.arguments || [],TOTAL_MEMORY: message.TOTAL_MEMORY || false};postMessage({"type" : "start","data" : Module.arguments.join(" ")});postMessage({"type" : "stdout","data" : "Received command: " +Module.arguments.join(" ") +((Module.TOTAL_MEMORY) ? ".  Processing with " + Module.TOTAL_MEMORY + " bits." : "")});var time = now();var result = ffmpeg_run(Module);var totalTime = now() - time;postMessage({"type" : "stdout","data" : "Finished processing (took " + totalTime + "ms)"});postMessage({"type" : "done","data" : result,"time" : totalTime});}};postMessage({"type" : "ready"});'], {
@@ -166,7 +167,7 @@ <h2 class="header">Record Video and convert into "mp4" using <a href="https://go
                     worker.onmessage = function(event) {
                         var message = event.data;
                         if (message.type == "ready") {
-                            log('<a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file has been loaded.');
+                            log('<a href="'+ workerPath +'" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file has been loaded.');
 
                             workerReady = true;
                             if (buffersReady)
@@ -174,7 +175,7 @@ <h2 class="header">Record Video and convert into "mp4" using <a href="https://go
                         } else if (message.type == "stdout") {
                             log(message.data);
                         } else if (message.type == "start") {
-                            log('<a href="https://googledrive.com/host/0B6GWd_dUUTT8OEtLRGdQb2pibDg/ffmpeg_asm.js" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file received ffmpeg command.');
+                            log('<a href="'+ workerPath +'" download="ffmpeg-asm.js">ffmpeg-asm.js</a> file received ffmpeg command.');
                         } else if (message.type == "done") {
                             log(JSON.stringify(message));
 
@@ -195,7 +196,12 @@ <h2 class="header">Record Video and convert into "mp4" using <a href="https://go
 
                         worker.postMessage({
                             type: 'command',
-                            arguments: ['-i', videoFile, '-c:v', 'mpeg4', '-b:v', '64k', '-strict', 'experimental', 'output.mp4'],
+                            arguments: [
+                                '-i', videoFile, 
+                                '-c:v', 'mpeg4', 
+                                '-b:v', '6400k', 
+                                '-strict', 'experimental', 'output.mp4'
+                            ],
                             files: [
                                 {
                                     data: new Uint8Array(aab),
@@ -281,6 +287,7 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
+        <script src="https://cdn.webrtc-experiment.com/ffmpeg_asm.js" async> </script>
     </body>
 </html>
diff --git a/getMediaElement/README.md b/getMediaElement/README.md
index 1b377614..e1495d80 100644
--- a/getMediaElement/README.md
+++ b/getMediaElement/README.md
@@ -1,12 +1,12 @@
 #### getMediaElement.js: A reusable library for all WebRTC applications! / [Demo](https://www.webrtc-experiment.com/getMediaElement/)
 
 ```html
-<script src="//www.webrtc-experiment.com/getMediaElement-v1.2.js"></script>
+<script src="//cdn.webrtc-experiment.com/getMediaElement.js"></script>
 ```
 
 This library generates HTML Audio/Video element with rich user-interface and advance media controls. It gives you full control over each control button; and its functionality!
 
-<img src="https://www.webrtc-experiment.com/images/getMediaElement.js.gif" />
+<img src="//cdn.webrtc-experiment.com/images/getMediaElement.js.gif" />
 
 ```javascript
 // you can pass HTML Video Element
diff --git a/navigator.customGetUserMediaBar/README.md b/navigator.customGetUserMediaBar/README.md
index 73cbff94..abfb7142 100644
--- a/navigator.customGetUserMediaBar/README.md
+++ b/navigator.customGetUserMediaBar/README.md
@@ -1,13 +1,13 @@
 ## [navigator.customGetUserMediaBar.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/navigator.customGetUserMediaBar) : Keep Users Privacy! / [Demo](https://www.WebRTC-Experiment.com/navigator.customGetUserMediaBar/)
 
-<img src="https://www.webrtc-experiment.com/images/navigator.customGetUserMediaBar.gif" style="width:100%;" />
+<img src="//cdn.webrtc-experiment.com/images/navigator.customGetUserMediaBar.gif" style="width:100%;" />
 
 =
 
 ### How to use it?
 
 ```html
-<script src="//www.webrtc-experiment.com/navigator.customGetUserMediaBar.js"></script>
+<script src="//cdn.webrtc-experiment.com/navigator.customGetUserMediaBar.js"></script>
 ```
 
 =
diff --git a/navigator.customGetUserMediaBar/index.html b/navigator.customGetUserMediaBar/index.html
index 37144977..d970b87a 100644
--- a/navigator.customGetUserMediaBar/index.html
+++ b/navigator.customGetUserMediaBar/index.html
@@ -14,7 +14,7 @@
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
         
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio {
@@ -54,7 +54,7 @@
         </script>
         
         <!-- script used to display custom notification bar! -->
-        <script src="https://www.webrtc-experiment.com/navigator.customGetUserMediaBar.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/navigator.customGetUserMediaBar.js"> </script>
     </head>
 
     <body>
@@ -87,7 +87,7 @@ <h1>
             <div class="github-stargazers"></div>
             
             <section class="experiment">
-                <img src="https://www.webrtc-experiment.com/images/navigator.customGetUserMediaBar.gif" style="width:100%;" />
+                <img src="//cdn.webrtc-experiment.com/images/navigator.customGetUserMediaBar.gif" style="width:100%;" />
                 <button id="invoke-customGetUserMediaBar" style="font-size:2em;">Invoke navigator.customGetUserMediaBar</button>
             </section>
             
@@ -141,7 +141,7 @@ <h1>
                 <h2 class="header">
                     How to use <a href="https://github.com/muaz-khan/WebRTC-Experiment/tree/master/navigator.customGetUserMediaBar" target="_blank">navigator.customGetUserMediaBar</a>?</h2>
                 <pre>
-&lt;script src="https://www.WebRTC-Experiment.com/navigator.customGetUserMediaBar.js"&gt;&lt;/script&gt;
+&lt;script src="//cdn.WebRTC-Experiment.com/navigator.customGetUserMediaBar.js"&gt;&lt;/script&gt;
 </pre>
             </section>
             <section class="experiment">
@@ -201,6 +201,6 @@ <h2 class="header" id="updates" style="color: red; padding-bottom: .1em;"><a hre
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/part-of-screen-sharing/README.md b/part-of-screen-sharing/README.md
index 4b8f1dbd..06d34cb2 100644
--- a/part-of-screen-sharing/README.md
+++ b/part-of-screen-sharing/README.md
@@ -10,10 +10,10 @@ Files:
 * [screenshot.js](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/part-of-screen-sharing/screenshot.js) � minified
 
 ```html
-<script src="//www.webrtc-experiment.com/screenshot.js"></script>
+<script src="//cdn.webrtc-experiment.com/screenshot.js"></script>
 
 <!-- or Developers version -->
-<script src="//www.webrtc-experiment.com/screenshot-dev.js"></script>
+<script src="//cdn.webrtc-experiment.com/screenshot-dev.js"></script>
 ```
 
 Remember, `grabMouse` is used by default. You need to make it `false` to make sure your resulting screenshot has no cursor icon.
diff --git a/video-conferencing/README.md b/video-conferencing/README.md
index 0eba3735..21508244 100644
--- a/video-conferencing/README.md
+++ b/video-conferencing/README.md
@@ -17,9 +17,9 @@
 #### Want to use video-conferencing in your own webpage?
 
 ```html
-<script src="//www.webrtc-experiment.com/socket.io.js"> </script>
-<script src="//www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"> </script>
-<script src="//www.webrtc-experiment.com/video-conferencing/conference.js"> </script>
+<script src="//cdn.webrtc-experiment.com/socket.io.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCPeerConnection-v1.5.js"> </script>
+<script src="//cdn.webrtc-experiment.com/video-conferencing/conference.js"> </script>
 
 <button id="setup-new-room">Setup New Conference</button>
 <table style="width: 100%;" id="rooms-list"></table>
@@ -32,8 +32,8 @@ var config = {
         // http://socketio-over-nodejs.nodejitsu.com:80/
         // http://webrtc-signaling.nodejitsu.com:80/
 
-        var SIGNALING_SERVER = 'http://webrtc-signaling.nodejitsu.com:80/',
-            defaultChannel = location.hash.substr(1) || location.href.replace(/\/|:|#|%|\.|\[|\]/g, '');
+        var SIGNALING_SERVER = 'https://webrtc-signaling.nodejitsu.com:443/',
+            defaultChannel = location.href.replace(/\/|:|#|%|\.|\[|\]/g, '');
 
         var channel = config.channel || defaultChannel;
         var sender = Math.round(Math.random() * 999999999) + 999999999;
diff --git a/video-conferencing/index.html b/video-conferencing/index.html
index 8b92d0ae..7edbae7a 100644
--- a/video-conferencing/index.html
+++ b/video-conferencing/index.html
@@ -13,7 +13,8 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio, video {
@@ -58,13 +59,12 @@
         </script>
         
         <!-- scripts used for video-conferencing -->
-        <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/video-conferencing/conference.js"> </script>
-        <script src="https://www.webrtc-experiment.com/getMediaElement.min.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/RTCPeerConnection-v1.5.js"> </script>
+        <script src="//cdn.webrtc-experiment.com/video-conferencing/conference.js"> </script>
         
-        <!-- script used to display custom notification bar! -->
-        <script src="https://www.webrtc-experiment.com/navigator.customGetUserMediaBar.js"> </script>
+        <!-- script used to stylize video element -->
+        <script src="//cdn.webrtc-experiment.com/getMediaElement.min.js"> </script>
     </head>
 
     <body>
@@ -202,30 +202,7 @@ <h1>
                     });
                 }
 
-                function captureUserMedia(callback, failure_callback, forceSkipCustomGetUserMediaBar) {
-                    if(!forceSkipCustomGetUserMediaBar) {
-                        var mediaConstraints = {
-                            audio: true,
-                            video: true
-                        };
-
-                        navigator.customGetUserMediaBar(mediaConstraints, function () {
-
-                            // now you can invoke "getUserMedia" to seamlessly capture user's media
-                            captureUserMedia(callback, failure_callback, true);
-
-                        }, function () {
-
-                            // user clicked "Deny" or "x" button
-                            // throw new Error('PermissionDeniedError');
-                            alert('PermissionDeniedError: User denied permission.');
-                            
-                            if(failure_callback) failure_callback();
-
-                        });
-                        return;
-                    }
-
+                function captureUserMedia(callback, failure_callback) {
                     var video = document.createElement('video');
 
                     getUserMedia({
@@ -321,9 +298,9 @@ <h1>
             <section class="experiment">
                 <h2 class="header">Want to use video-conferencing in your own webpage?</h2>
                 <pre>
-&lt;script src="//www.webrtc-experiment.com/socket.io.js"&gt; &lt;/script&gt;
-&lt;script src="//www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"&gt; &lt;/script&gt;
-&lt;script src="//www.webrtc-experiment.com/video-conferencing/conference.js"&gt; &lt;/script&gt;
+&lt;script src="//cdn.webrtc-experiment.com/socket.io.js"&gt; &lt;/script&gt;
+&lt;script src="//cdn.webrtc-experiment.com/RTCPeerConnection-v1.5.js"&gt; &lt;/script&gt;
+&lt;script src="//cdn.webrtc-experiment.com/video-conferencing/conference.js"&gt; &lt;/script&gt;
 
 &lt;button id="setup-new-room"&gt;Setup New Conference&lt;/button&gt;
 &lt;table style="width: 100%;" id="rooms-list"&gt;&lt;/table&gt;
@@ -336,8 +313,8 @@ <h2 class="header">Want to use video-conferencing in your own webpage?</h2>
         // http://socketio-over-nodejs.nodejitsu.com:80/
         // http://webrtc-signaling.nodejitsu.com:80/
 
-        var SIGNALING_SERVER = 'http://webrtc-signaling.nodejitsu.com:80/',
-            defaultChannel = location.hash.substr(1) || location.href.replace(/\/|:|#|%|\.|\[|\]/g, '');
+        var SIGNALING_SERVER = 'https://webrtc-signaling.nodejitsu.com:443/',
+            defaultChannel = location.href.replace(/\/|:|#|%|\.|\[|\]/g, '');
 
         var channel = config.channel || defaultChannel;
         var sender = Math.round(Math.random() * 999999999) + 999999999;
@@ -457,6 +434,6 @@ <h2 class="header" id="feedback">Feedback</h2>
         </footer>
     
         <!-- commits.js is useless for you! -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>
diff --git a/webrtc-broadcasting/README.md b/webrtc-broadcasting/README.md
index 72726546..cbc737aa 100644
--- a/webrtc-broadcasting/README.md
+++ b/webrtc-broadcasting/README.md
@@ -123,9 +123,9 @@ All these three pages has unique channels. They'll NEVER see rooms from each oth
 #### Want to use video-conferencing in your own webpage?
 
 ```html
-<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="//cdn.webrtc-experiment.com/socket.io.js"> </script>
+<script src="//cdn.webrtc-experiment.com/RTCPeerConnection-v1.5.js"> </script>
+<script src="//cdn.webrtc-experiment.com/webrtc-broadcasting/broadcast.js"> </script>
 
 <select id="broadcasting-option">
     <option>Audio + Video</option>
diff --git a/webrtc-broadcasting/index.html b/webrtc-broadcasting/index.html
index d28201be..912a36e4 100644
--- a/webrtc-broadcasting/index.html
+++ b/webrtc-broadcasting/index.html
@@ -14,7 +14,8 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
-        <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
+        
+        <link rel="stylesheet" href="//cdn.webrtc-experiment.com/style.css">
         
         <style>
             audio, video {
@@ -71,12 +72,12 @@
         </script>
         
         <!-- scripts used for broadcasting -->
-        <script src="https://www.webrtc-experiment.com/firebase.js"> </script> <!-- Signaling -->
-        <script src="https://www.webrtc-experiment.com/RTCPeerConnection-v1.5.js"> </script> <!-- WebRTC simple wrapper -->
-        <script src="https://www.webrtc-experiment.com/webrtc-broadcasting/broadcast.js"> </script> <!-- Multi-user connectivity handler -->
+        <script src="//cdn.webrtc-experiment.com/firebase.js"> </script> <!-- Signaling -->
+        <script src="//cdn.webrtc-experiment.com/RTCPeerConnection-v1.5.js"> </script> <!-- WebRTC simple wrapper -->
+        <script src="//cdn.webrtc-experiment.com/webrtc-broadcasting/broadcast.js"> </script> <!-- Multi-user connectivity handler -->
         
         <!-- This Library is used to detect WebRTC features -->
-        <script src="https://www.webrtc-experiment.com/DetectRTC.js"></script>
+        <script src="//cdn.webrtc-experiment.com/DetectRTC.js"></script>
     </head>
 
     <body>
@@ -379,6 +380,6 @@ <h2 class="header">How it works?</h2>
         </footer>
     
         <!-- commits.js is useless for you! It is not part of this WebRTC Experiment. -->
-        <script src="https://www.webrtc-experiment.com/commits.js" async> </script>
+        <script src="//cdn.webrtc-experiment.com/commits.js" async> </script>
     </body>
 </html>