diff --git a/DataChannel/DataChannel.js b/DataChannel/DataChannel.js index 50582499..bf303af0 100644 --- a/DataChannel/DataChannel.js +++ b/DataChannel/DataChannel.js @@ -93,19 +93,23 @@ if (self.config) return; self.config = { - onroom: function(room) { + ondatachannel: function(room) { if (!dataConnector) { self.room = room; return; } + var tempRoom = { + id: room.roomToken, + owner: room.broadcaster + }; + + if (self.ondatachannel) return self.ondatachannel(tempRoom); + if (self.joinedARoom) return; self.joinedARoom = true; - dataConnector.joinRoom({ - roomToken: room.roomToken, - joinUser: room.broadcaster - }); + self.join(tempRoom); }, onopen: function(userid, _channel) { self.onopen(userid, _channel); @@ -153,7 +157,7 @@ fileReceiver = new FileReceiver(); textReceiver = new TextReceiver(); - if (self.room) self.config.onroom(self.room); + if (self.room) self.config.ondatachannel(self.room); } this.open = function(_channel) { @@ -175,6 +179,18 @@ prepareInit(init); }; + // manually join a room + this.join = function(room) { + if (!room.id || !room.owner) { + throw 'Invalid room info passed.'; + } + + dataConnector.joinRoom({ + roomToken: room.id, + joinUser: room.owner + }); + }; + this.send = function(data, _channel) { if (!data) throw 'No file, data or text message to share.'; if (data.size) @@ -515,7 +531,7 @@ onmessage: function(response) { if (response.userToken == self.userToken) return; - if (isGetNewRoom && response.roomToken && response.broadcaster) config.onroom(response); + if (isGetNewRoom && response.roomToken && response.broadcaster) config.ondatachannel(response); if (response.newParticipant) onNewParticipant(response.newParticipant); @@ -627,7 +643,7 @@ packets = 0; // uuid is used to uniquely identify sending instance - var uuid = getRandomString(); + file.uuid = getRandomString(); var reader = new window.FileReader(); reader.readAsDataURL(file); @@ -636,7 +652,7 @@ function onReadAsDataURL(event, text) { var data = { type: 'file', - uuid: uuid + uuid: file.uuid }; if (event) { @@ -649,7 +665,7 @@ remaining: packets--, length: numberOfPackets, sent: numberOfPackets - packets - }, uuid); + }, file.uuid); if (text.length > packetSize) data.message = text.slice(0, packetSize); else { @@ -657,7 +673,7 @@ data.last = true; data.name = file.name; - if (config.onFileSent) config.onFileSent(file, uuid); + if (config.onFileSent) config.onFileSent(file, file.uuid); } // WebRTC-DataChannels.send(data, privateDataChannel) @@ -709,7 +725,7 @@ // if you don't want to auto-save to disk: // channel.autoSaveToDisk=false; if (root.autoSaveToDisk) - FileSaver.SaveToDisk(virtualURL, data.name); + FileSaver.SaveToDisk(dataURL, data.name); // channel.onFileReceived = function(fileName, file) {} // file.blob || file.dataURL || file.url || file.uuid @@ -885,7 +901,7 @@ credential: 'homeo', username: 'homeo' }; - iceServers.iceServers = [TURN, STUN]; + iceServers.iceServers = [STUN, TURN]; } var optional = { diff --git a/DataChannel/README.md b/DataChannel/README.md index fd80fb55..9b7117f2 100644 --- a/DataChannel/README.md +++ b/DataChannel/README.md @@ -89,6 +89,27 @@ channel.onmessage = function(message, userid, latency) { } = +##### `ondatachannel` + +Allows you show list of all available data channels to the user; and let him choose which one to join: + +```javascript +channel.ondatachannel = function(data_channel) { + channel.join(data_channel); + + // or + channel.join({ + id: data_channel.id, + owner: data_channel.owner + }); + + // id: unique identifier for the session + // owner: unique identifier for the session initiator +}; +``` + += + ##### Use custom user-ids ```javascript @@ -160,6 +181,17 @@ channel.leave(); // closing entire session = +##### `uuid` for files + +You can get `uuid` for each file (being sent) like this: + +```javascript +channel.send(file); +var uuid = file.uuid; // "file"-Dot-uuid +``` + += + ##### To Share files ```javascript diff --git a/DataChannel/auto-session-establishment.html b/DataChannel/auto-session-establishment.html index 45269bfa..444fccb2 100644 --- a/DataChannel/auto-session-establishment.html +++ b/DataChannel/auto-session-establishment.html @@ -298,8 +298,8 @@

Share Files

+ '
' + data + '
'; } - if (!parent) chatOutput.insertBefore(div, chatOutput.firstChild); - else parent.insertBefore(div, parent.firstChild); + if (!parent) chatOutput.appendChild(div, chatOutput.firstChild); + else fileProgress.appendChild(div, fileProgress.firstChild); div.tabIndex = 0; div.focus(); diff --git a/DataChannel/index.html b/DataChannel/index.html index 2e81632f..83aae0cc 100644 --- a/DataChannel/index.html +++ b/DataChannel/index.html @@ -1,452 +1,323 @@ - + + + + DataChannel.js » A WebRTC Library for Data Sharing ® Muaz Khan + + + + + + + + + + + + + + + + +
+
+

+ DataChannel.js + » A WebRTC Library for Data Sharing +

+

+ Copyright © 2013 + Muaz Khan<@muazkh> + » + @WebRTC Experiments + » + Google+ + » + What's New? +

+
+ +
+ +
+
+
+ + +
+
+

Open New DataChannel Connection

+ +
+ +
+ + + + + +
+

Text Chat

+ +
+ + 		+

Share Files

+ + +
+
+
+ + + +
+

Latest Updates

+
+
+ +
+

Feedback

+
+ +
+ Enter your email too; if you want "direct" reply! +
+ +
+

DataChannel.js Features:

+
    +
  1. Direct messages — to any user using his `user-id`
    2. +
  2. Eject/Reject any user — using his `user-id`
    3. +
  3. Leave any room (i.e. data session) or close entire session using `leave` method
    4. +
  4. File size is limitless!
    5. +
  5. Text message length is limitless!
    6. +
  6. Size of data is also limitless!
    7. +
  7. Fallback to firebase/socket.io/websockets/etc.
    8. +
  8. Users' presence detection using `onleave`
    9. +
  9. Latency detection
    10. +
  10. Multi-longest strings/files concurrent transmission
    11. +
+
+ +
+

How to use DataChannel.js?

+ 
+<script src="https://www.webrtc-experiment.com/DataChannel.js"> </script>
+
+<input type="text" id="chat-input" disabled 
+       style="font-size: 2em; width: 98%;"><br />
+<div id="chat-output"></div>
 
-
-    File Sharing + Text Chat using WebRTC DataChannel ® Muaz Khan
-    
-    
-    
-    
-    
-    
-    
-    
-    
-
-    
-    
-
-
-
-

-↑
-    WEBRTC EXPERIMENTS
-
-
-
File Sharing + Text Chat using WebRTC DataChannel
-

-
-

-    Copyright © 2013
-    Muaz Khan<@muazkh>.
-

-

-    

-

-

-    
Open Data Channel

-    
-    
-    
or join:

-    
-

-
-    
-        
-        
-    
-

-            
Text Chat

-
-            

-            
-        		
-            
Share Files

-            
-
-            

-        

-
-
-
-
-

-

-
-
Getting started with WebRTC DataChannel

-            
-<script src="https://www.webrtc-experiment.com/DataChannel.js"></script>
-<script>
-    var channel = new DataChannel();
-
-    // to create/open a new channel
-    channel.open('channel-name');
-
-    // to send text/data or file
-    channel.send(file || data || 'text');
-	
-    // if soemone already created a channel; to join it: use "connect" method
-    channel.connect('channel-name');
 </script>
-

-Remember, A-to-Z, everything is optional! You can set channel-name in constructor or in
-open/connect methods. It is your choice!
-
-

-

-
-
Features:

-
    
-    
  1. Send file directly — of any size
    2. 
-    
  2. Send text-message of any length
    3. 
-    
  3. Send data directly
    4. 
-    
  4. Simplest syntax ever! Same like WebSockets.
    5. 
-    
  5. Supports fallback to socket.io/websockets/etc.
    6. 
-    
  6. Auto users' presence detection
    7. 
-    
  7. Allows you eject any user; or close your entire data session
    8. 
-

-

-

-
-
Additional:

-            
+
 <script>
-    // to be alerted on data ports get open
-    channel.onopen = function(userid) {}
-	
-    // to be alerted on data ports get new message
-    channel.onmessage = function(message, userid) {}
-	
-    // by default; connection is [many-to-many]; you can use following directions
-    channel.direction = 'one-to-one';
-    channel.direction = 'one-to-many';
-    channel.direction = 'many-to-many';	// --- it is default
-
-    // show progress bar!
-    channel.onFileProgress = function (packets) {
-        // packets.remaining
-        // packets.sent
-        // packets.received
-        // packets.length
+    var channel = new DataChannel('Session Unique Identifier');
+
+    channel.onopen = function(userid) {
+        chatInput.disabled = false;
+        chatInput.value = 'Hi, ' + userid;
+        chatInput.focus();
     };
 
-    // on file successfully sent
-    channel.onFileSent = function (file) {
-        // file.name
-        // file.size
+    channel.onmessage = function(message, userid) {
+        chatOutput.innerHTML = userid + ': ' + message + '<hr />' 
+                             + chatOutput.innerHTML;
     };
 
-    // on file successfully received
-    channel.onFileReceived = function (fileName) {};
+    channel.onleave = function(userid) {
+        chatOutput.innerHTML = userid + ' Left.<hr />' 
+                             + chatOutput.innerHTML;
+    };
 </script>
 

-

-

-
-
Errors Handling

-            
-<script>
-    // error to open data ports
-    channel.onerror = function(event) {}
+
+
+ - // data ports suddenly dropped - channel.onclose = function(event) {} -</script> - -
-
- -

Use your own socket.io for signaling

- 
-<script>
-    // by default Firebase is used for signaling; you can override it
-    channel.openSignalingChannel = function(config) {
-        var socket = io.connect('http://your-site:8888');
-        socket.channel = config.channel || this.channel || 'default-channel';
-        socket.on('message', config.onmessage);
-
-        socket.send = function (data) {
-            socket.emit('message', data);
-        };
-
-        if (config.onopen) setTimeout(config.onopen, 1);
-        return socket;
-    }
-</script>
-
-
-
-
-

Feedback

- -
- -
- -
- - - - - \ No newline at end of file + + + + diff --git a/DataChannel/simple.html b/DataChannel/simple.html new file mode 100644 index 00000000..c9f3d158 --- /dev/null +++ b/DataChannel/simple.html @@ -0,0 +1,452 @@ + + + + + File Sharing + Text Chat using WebRTC DataChannel ® Muaz Khan + + + + + + + + + + + + + + + +
+↑ + WEBRTC EXPERIMENTS + + +

File Sharing + Text Chat using WebRTC DataChannel +

+ +

+ Copyright © 2013 + Muaz Khan<@muazkh>. +

+
+
+
+
+

Open Data Channel

+ + +

or join:

+ +
+ + + + + +
+

Text Chat

+ +
+ + 		+

Share Files

+ + +
+
+ + + + +
+
+ +

Getting started with WebRTC DataChannel

+ 
+<script src="https://www.webrtc-experiment.com/DataChannel.js"></script>
+<script>
+    var channel = new DataChannel();
+
+    // to create/open a new channel
+    channel.open('channel-name');
+
+    // to send text/data or file
+    channel.send(file || data || 'text');
+	
+    // if soemone already created a channel; to join it: use "connect" method
+    channel.connect('channel-name');
+</script>
+
+Remember, A-to-Z, everything is optional! You can set channel-name in constructor or in +open/connect methods. It is your choice! + +
+
+ +

Features:

+
    +
  1. Send file directly — of any size
    2. +
  2. Send text-message of any length
    3. +
  3. Send data directly
    4. +
  4. Simplest syntax ever! Same like WebSockets.
    5. +
  5. Supports fallback to socket.io/websockets/etc.
    6. +
  6. Auto users' presence detection
    7. +
  7. Allows you eject any user; or close your entire data session
    8. +
+
+
+ +

Additional:

+ 
+<script>
+    // to be alerted on data ports get open
+    channel.onopen = function(userid) {}
+	
+    // to be alerted on data ports get new message
+    channel.onmessage = function(message, userid) {}
+	
+    // by default; connection is [many-to-many]; you can use following directions
+    channel.direction = 'one-to-one';
+    channel.direction = 'one-to-many';
+    channel.direction = 'many-to-many';	// --- it is default
+
+    // show progress bar!
+    channel.onFileProgress = function (packets) {
+        // packets.remaining
+        // packets.sent
+        // packets.received
+        // packets.length
+    };
+
+    // on file successfully sent
+    channel.onFileSent = function (file) {
+        // file.name
+        // file.size
+    };
+
+    // on file successfully received
+    channel.onFileReceived = function (fileName) {};
+</script>
+
+
+
+ +

Errors Handling

+ 
+<script>
+    // error to open data ports
+    channel.onerror = function(event) {}
+	
+    // data ports suddenly dropped
+    channel.onclose = function(event) {}
+</script>
+
+
+
+ +

Use your own socket.io for signaling

+ 
+<script>
+    // by default Firebase is used for signaling; you can override it
+    channel.openSignalingChannel = function(config) {
+        var socket = io.connect('http://your-site:8888');
+        socket.channel = config.channel || this.channel || 'default-channel';
+        socket.on('message', config.onmessage);
+
+        socket.send = function (data) {
+            socket.emit('message', data);
+        };
+
+        if (config.onopen) setTimeout(config.onopen, 1);
+        return socket;
+    }
+</script>
+
+
+
+
+

Feedback

+ +
+ +
+ +
+
+ + + + \ No newline at end of file diff --git a/Pluginfree-Screen-Sharing/README.md b/Pluginfree-Screen-Sharing/README.md index c52e986c..d3b9a649 100644 --- a/Pluginfree-Screen-Sharing/README.md +++ b/Pluginfree-Screen-Sharing/README.md @@ -31,7 +31,8 @@ There is another experiment: [WebRTC Tab Sharing using experimental tabCapture A It is a big wish to share desktop using RTCWeb peer connection APIs but unfortunately currently it is not possible. -Current experiment is using chrome screen sharing APIs which is allows end-users just **view the screen**....nothing else! +Current experiment is using chrome screen sharing APIs (media/constraints) which allows end-users to **view the screen**....nothing else! + = #### Browser Support diff --git a/video-conferencing/conference-ui.js b/Pluginfree-Screen-Sharing/conference-ui.js similarity index 62% rename from video-conferencing/conference-ui.js rename to Pluginfree-Screen-Sharing/conference-ui.js index b3ff657c..f8ae4f19 100644 --- a/video-conferencing/conference-ui.js +++ b/Pluginfree-Screen-Sharing/conference-ui.js @@ -1,11 +1,11 @@ // 2013, @muazkh » github.com/muaz-khan // MIT License » https://webrtc-experiment.appspot.com/licence/ -// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing +// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing var config = { - openSocket: function(config) { + openSocket: function (config) { var SIGNALING_SERVER = 'https://www.webrtc-experiment.com:8553/', - defaultChannel = location.hash.substr(1) || 'video-conferencing-hangout'; + defaultChannel = location.hash.substr(1) || 'Pluginfree-Screen-Sharing'; var channel = config.channel || defaultChannel; var sender = Math.round(Math.random() * 999999999) + 999999999; @@ -30,22 +30,14 @@ var config = { socket.on('message', config.onmessage); }, - onRemoteStream: function(media) { + onRemoteStream: function (media) { var video = media.video; - video.setAttribute('controls', true); - video.setAttribute('id', media.stream.id); - participants.insertBefore(video, participants.firstChild); - video.play(); rotateVideo(video); }, - onRemoteStreamEnded: function(stream) { - var video = document.getElementById(stream.id); - if (video) video.parentNode.removeChild(video); - }, - onRoomFound: function(room) { + onRoomFound: function (room) { var alreadyExist = document.getElementById(room.broadcaster); if (alreadyExist) return; @@ -54,26 +46,29 @@ var config = { var tr = document.createElement('tr'); tr.setAttribute('id', room.broadcaster); tr.innerHTML = '' + room.roomName + '' + - ''; + ''; roomsList.insertBefore(tr, roomsList.firstChild); - tr.onclick = function() { + tr.onclick = function () { var tr = this; - captureUserMedia(function() { - conferenceUI.joinRoom({ - roomToken: tr.querySelector('.join').id, - joinUser: tr.id - }); + conferenceUI.joinRoom({ + roomToken: tr.querySelector('.join').id, + joinUser: tr.id }); hideUnnecessaryStuff(); }; + }, + onNewParticipant: function (participants) { + var numberOfParticipants = document.getElementById('number-of-participants'); + if (!numberOfParticipants) return; + numberOfParticipants.innerHTML = participants + ' room participants'; } }; function createButtonClickHandler() { - captureUserMedia(function() { + captureUserMedia(function () { conferenceUI.createRoom({ - roomName: (document.getElementById('conference-name') || { }).value || 'Anonymous' + roomName: ((document.getElementById('conference-name') || {}).value || 'Anonymous') + ' shared screen with you' }); }); hideUnnecessaryStuff(); @@ -85,23 +80,37 @@ function captureUserMedia(callback) { video.setAttribute('controls', true); participants.insertBefore(video, participants.firstChild); + var screen_constraints = { + mandatory: { + chromeMediaSource: 'screen' + }, + optional: [] + }; + var constraints = { + audio: false, + video: screen_constraints + }; getUserMedia({ video: video, - onsuccess: function(stream) { + constraints: constraints, + onsuccess: function (stream) { config.attachStream = stream; callback && callback(); video.setAttribute('muted', true); rotateVideo(video); }, - onerror: function() { - alert('unable to get access to your webcam'); - callback && callback(); + onerror: function () { + if (location.protocol === 'http:') { + alert('Please test this WebRTC experiment on HTTPS.'); + } else { + alert('Screen capturing is either denied or not supported. Are you enabled flag: "Enable screen capture support in getUserMedia"?'); + } } }); } -// You can use! window.onload = function() {} +/* on page load: get public rooms */ var conferenceUI = conference(config); /* UI specific */ @@ -121,14 +130,13 @@ function hideUnnecessaryStuff() { function rotateVideo(video) { video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(0deg)'; - setTimeout(function() { + setTimeout(function () { video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(360deg)'; }, 1000); } -(function() { +(function () { var uniqueToken = document.getElementById('unique-token'); - if (uniqueToken) - if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '

Share this link

'; - else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace( /\./g , '-'); -})(); + if (uniqueToken) if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '

Share this link

'; + else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace(/\./g, '-'); +})(); \ No newline at end of file diff --git a/Pluginfree-Screen-Sharing/conference.js b/Pluginfree-Screen-Sharing/conference.js new file mode 100644 index 00000000..03462129 --- /dev/null +++ b/Pluginfree-Screen-Sharing/conference.js @@ -0,0 +1,241 @@ +/* MIT License: https://webrtc-experiment.appspot.com/licence/ */ + +var conference = function (config) { + var self = { + userToken: uniqueToken() + }, + channels = '--', + isbroadcaster, + isGetNewRoom = true, + participants = 1, + defaultSocket = {}; + + function openDefaultSocket() { + defaultSocket = config.openSocket({ + onmessage: defaultSocketResponse, + callback: function (socket) { + defaultSocket = socket; + } + }); + } + + function defaultSocketResponse(response) { + if (response.userToken == self.userToken) return; + + if (isGetNewRoom && response.roomToken && response.broadcaster) config.onRoomFound(response); + + if (response.newParticipant) onNewParticipant(response.newParticipant); + + if (response.userToken && response.joinUser == self.userToken && response.participant && channels.indexOf(response.userToken) == -1) { + channels += response.userToken + '--'; + openSubSocket({ + isofferer: true, + channel: response.channel || response.userToken, + closeSocket: true + }); + } + } + + function openSubSocket(_config) { + if (!_config.channel) return; + var socketConfig = { + channel: _config.channel, + onmessage: socketResponse, + onopen: function () { + if (isofferer && !peer) initPeer(); + } + }; + + socketConfig.callback = function (_socket) { + socket = _socket; + this.onopen(); + }; + + var socket = config.openSocket(socketConfig), + isofferer = _config.isofferer, + gotstream, + htmlElement = document.createElement('video'), + inner = {}, + peer; + + var peerConfig = { + attachStream: config.attachStream, + onICE: function (candidate) { + socket && socket.send({ + userToken: self.userToken, + candidate: { + sdpMLineIndex: candidate.sdpMLineIndex, + candidate: JSON.stringify(candidate.candidate) + } + }); + }, + onRemoteStream: function (stream) { + htmlElement[moz ? 'mozSrcObject' : 'src'] = moz ? stream : webkitURL.createObjectURL(stream); + htmlElement.play(); + + _config.stream = stream; + onRemoteStreamStartsFlowing(); + } + }; + + function initPeer(offerSDP) { + if (!offerSDP) peerConfig.onOfferSDP = sendsdp; + else { + peerConfig.offerSDP = offerSDP; + peerConfig.onAnswerSDP = sendsdp; + } + peer = RTCPeerConnection(peerConfig); + } + + function onRemoteStreamStartsFlowing() { + if (!(htmlElement.readyState <= HTMLMediaElement.HAVE_CURRENT_DATA || htmlElement.paused || htmlElement.currentTime <= 0)) { + afterRemoteStreamStartedFlowing(); + } else setTimeout(onRemoteStreamStartsFlowing, 50); + } + + function afterRemoteStreamStartedFlowing() { + gotstream = true; + + config.onRemoteStream({ + video: htmlElement + }); + + if (isbroadcaster && channels.split('--').length > 3) { + /* broadcasting newly connected participant for video-conferencing! */ + defaultSocket && defaultSocket.send({ + newParticipant: socket.channel, + userToken: self.userToken + }); + } + + /* closing subsocket here on the offerer side */ + if (_config.closeSocket) socket = null; + } + + function sendsdp(sdp) { + sdp = JSON.stringify(sdp); + var part = parseInt(sdp.length / 3); + + var firstPart = sdp.slice(0, part), + secondPart = sdp.slice(part, sdp.length - 1), + thirdPart = ''; + + if (sdp.length > part + part) { + secondPart = sdp.slice(part, part + part); + thirdPart = sdp.slice(part + part, sdp.length); + } + + socket && socket.send({ + userToken: self.userToken, + firstPart: firstPart + }); + + socket && socket.send({ + userToken: self.userToken, + secondPart: secondPart + }); + + socket && socket.send({ + userToken: self.userToken, + thirdPart: thirdPart + }); + } + + function socketResponse(response) { + if (response.userToken == self.userToken) return; + if (response.firstPart || response.secondPart || response.thirdPart) { + if (response.firstPart) { + inner.firstPart = response.firstPart; + if (inner.secondPart && inner.thirdPart) selfInvoker(); + } + if (response.secondPart) { + inner.secondPart = response.secondPart; + if (inner.firstPart && inner.thirdPart) selfInvoker(); + } + + if (response.thirdPart) { + inner.thirdPart = response.thirdPart; + if (inner.firstPart && inner.secondPart) selfInvoker(); + } + } + + if (response.candidate && !gotstream) { + peer && peer.addICE({ + sdpMLineIndex: response.candidate.sdpMLineIndex, + candidate: JSON.parse(response.candidate.candidate) + }); + } + } + + var invokedOnce = false; + + function selfInvoker() { + if (invokedOnce) return; + + invokedOnce = true; + + inner.sdp = JSON.parse(inner.firstPart + inner.secondPart + inner.thirdPart); + if (isofferer) { + peer.addAnswerSDP(inner.sdp); + if (config.onNewParticipant) config.onNewParticipant(participants++); + } else initPeer(inner.sdp); + } + } + + function startBroadcasting() { + defaultSocket && defaultSocket.send({ + roomToken: self.roomToken, + roomName: self.roomName, + broadcaster: self.userToken + }); + setTimeout(startBroadcasting, 3000); + } + + function onNewParticipant(channel) { + if (!channel || channels.indexOf(channel) != -1 || channel == self.userToken) return; + channels += channel + '--'; + + var new_channel = uniqueToken(); + openSubSocket({ + channel: new_channel, + closeSocket: true + }); + + defaultSocket.send({ + participant: true, + userToken: self.userToken, + joinUser: channel, + channel: new_channel + }); + } + + function uniqueToken() { + return Math.random().toString(36).substr(2, 35); + } + + openDefaultSocket(); + return { + createRoom: function (_config) { + self.roomName = _config.roomName || 'Anonymous'; + self.roomToken = uniqueToken(); + + isbroadcaster = true; + isGetNewRoom = false; + startBroadcasting(); + }, + joinRoom: function (_config) { + self.roomToken = _config.roomToken; + isGetNewRoom = false; + + openSubSocket({ + channel: self.userToken + }); + + defaultSocket.send({ + participant: true, + userToken: self.userToken, + joinUser: _config.joinUser + }); + } + }; +}; diff --git a/Pluginfree-Screen-Sharing/index.html b/Pluginfree-Screen-Sharing/index.html index 53a09a5c..c519916c 100644 --- a/Pluginfree-Screen-Sharing/index.html +++ b/Pluginfree-Screen-Sharing/index.html @@ -1,8 +1,303 @@ - -
- - -
- - - \ No newline at end of file + + + + + WebRTC Screen Sharing | Plugin-free ® Muaz Khan + + + + + + + + + + + + +
+ ↑ WEBRTC EXPERIMENTS + +

WebRTC Plugin-free Screen Sharing

+

+ Copyright © 2013 + Muaz Khan<@muazkh>. +

+ +
+
+
+ +
+ + + + + +
+ + + +
+ + + + +
Private sharing ?? +

+ /Pluginfree-Screen-Sharing/#123456789 +

+
+
+ + + + + +
+
+ +
+

Enable screen capture support in getUserMedia()

+ +

+ + Allow web pages to request access to the screen contents via the getUserMedia() API. #enable-usermedia-screen-capture + +

+ + You must enable this flag via "chrome://flags/" + +
+ +
+
+ +
+

To use code in your own site, you must understand following limitations:

+
+
+ Chrome Canary denies "screen capturing" request automatically if: +
    +
  1. You've not used 'chromeMediaSource' constraint: +
    + 
    +mandatory: {chromeMediaSource: 'screen'}
+
    + +
    2. +
  2. You requested audio-stream alongwith 'chromeMediaSource' – it is not permitted.
    3. +
  3. You're not testing it on SSL origin (HTTPS domain).
    4. +
  4. "screen capturing" is requested multiple times per tab. Maximum one request is permitted per page!
    5. +
+
+ Remember, recursive cascade images or blurred screen is chrome's implementation issues. It will be solved soon. +
+
+ mandatory: {chromeMediaSource: 'tab'} can only be useful in chrome extensions. See Tab sharing using tabCapture APIs. +
+ +
+
+ + It is preferred to use RTCMultiConnection.js for audio/video/screen conferencing/broadcasting. + +
+
+

Source Code on Github

+
+
+
+

Feedback

+
+ +
+ +
+
+ + + + diff --git a/Pluginfree-Screen-Sharing/screen-sharing.js b/Pluginfree-Screen-Sharing/screen-sharing.js deleted file mode 100644 index 00a316cc..00000000 --- a/Pluginfree-Screen-Sharing/screen-sharing.js +++ /dev/null @@ -1,673 +0,0 @@ -// 2013, @muazkh » github.com/muaz-khan -// MIT License » https://webrtc-experiment.appspot.com/licence/ -// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing - -var conference = function(config) { - var self = { - userToken: uniqueToken() - }, - channels = '--', - isbroadcaster, - isGetNewRoom = true, - participants = 1, - defaultSocket = { }; - - function openDefaultSocket() { - defaultSocket = config.openSocket({ - onmessage: defaultSocketResponse, - callback: function(socket) { - defaultSocket = socket; - } - }); - } - - function defaultSocketResponse(response) { - if (response.userToken == self.userToken) return; - - if (isGetNewRoom && response.roomToken && response.broadcaster) config.onRoomFound(response); - - if (response.newParticipant) onNewParticipant(response.newParticipant); - - if (response.userToken && response.joinUser == self.userToken && response.participant && channels.indexOf(response.userToken) == -1) { - channels += response.userToken + '--'; - openSubSocket({ - isofferer: true, - channel: response.channel || response.userToken, - closeSocket: true - }); - } - } - - function openSubSocket(_config) { - if (!_config.channel) return; - var socketConfig = { - channel: _config.channel, - onmessage: socketResponse, - onopen: function() { - if (isofferer && !peer) initPeer(); - } - }; - - socketConfig.callback = function(_socket) { - socket = _socket; - this.onopen(); - }; - - var socket = config.openSocket(socketConfig), - isofferer = _config.isofferer, - gotstream, - htmlElement = document.createElement('video'), - inner = { }, - peer; - - var peerConfig = { - attachStream: config.attachStream, - onICE: function(candidate) { - socket && socket.send({ - userToken: self.userToken, - candidate: { - sdpMLineIndex: candidate.sdpMLineIndex, - candidate: JSON.stringify(candidate.candidate) - } - }); - }, - onRemoteStream: function(stream) { - htmlElement[moz ? 'mozSrcObject' : 'src'] = moz ? stream : window.webkitURL.createObjectURL(stream); - htmlElement.play(); - - _config.stream = stream; - onRemoteStreamStartsFlowing(); - } - }; - - function initPeer(offerSDP) { - if (!offerSDP) peerConfig.onOfferSDP = sendsdp; - else { - peerConfig.offerSDP = offerSDP; - peerConfig.onAnswerSDP = sendsdp; - } - peer = RTCPeerConnection(peerConfig); - } - - function onRemoteStreamStartsFlowing() { - if (!(htmlElement.readyState <= HTMLMediaElement.HAVE_CURRENT_DATA || htmlElement.paused || htmlElement.currentTime <= 0)) { - afterRemoteStreamStartedFlowing(); - } else setTimeout(onRemoteStreamStartsFlowing, 50); - } - - function afterRemoteStreamStartedFlowing() { - gotstream = true; - - config.onRemoteStream({ - video: htmlElement - }); - - if (isbroadcaster && channels.split('--').length > 3) { - /* broadcasting newly connected participant for video-conferencing! */ - defaultSocket && defaultSocket.send({ - newParticipant: socket.channel, - userToken: self.userToken - }); - } - - /* closing subsocket here on the offerer side */ - if (_config.closeSocket) socket = null; - } - - function sendsdp(sdp) { - sdp = JSON.stringify(sdp); - var part = parseInt(sdp.length / 3); - - var firstPart = sdp.slice(0, part), - secondPart = sdp.slice(part, sdp.length - 1), - thirdPart = ''; - - if (sdp.length > part + part) { - secondPart = sdp.slice(part, part + part); - thirdPart = sdp.slice(part + part, sdp.length); - } - - socket && socket.send({ - userToken: self.userToken, - firstPart: firstPart - }); - - socket && socket.send({ - userToken: self.userToken, - secondPart: secondPart - }); - - socket && socket.send({ - userToken: self.userToken, - thirdPart: thirdPart - }); - } - - function socketResponse(response) { - if (response.userToken == self.userToken) return; - if (response.firstPart || response.secondPart || response.thirdPart) { - if (response.firstPart) { - inner.firstPart = response.firstPart; - if (inner.secondPart && inner.thirdPart) selfInvoker(); - } - if (response.secondPart) { - inner.secondPart = response.secondPart; - if (inner.firstPart && inner.thirdPart) selfInvoker(); - } - - if (response.thirdPart) { - inner.thirdPart = response.thirdPart; - if (inner.firstPart && inner.secondPart) selfInvoker(); - } - } - - if (response.candidate && !gotstream) { - peer && peer.addICE({ - sdpMLineIndex: response.candidate.sdpMLineIndex, - candidate: JSON.parse(response.candidate.candidate) - }); - } - } - - var invokedOnce = false; - - function selfInvoker() { - if (invokedOnce) return; - - invokedOnce = true; - - inner.sdp = JSON.parse(inner.firstPart + inner.secondPart + inner.thirdPart); - if (isofferer) { - peer.addAnswerSDP(inner.sdp); - if (config.onNewParticipant) config.onNewParticipant(participants++); - } else initPeer(inner.sdp); - } - } - - function startBroadcasting() { - defaultSocket && defaultSocket.send({ - roomToken: self.roomToken, - roomName: self.roomName, - broadcaster: self.userToken - }); - setTimeout(startBroadcasting, 3000); - } - - function onNewParticipant(channel) { - if (!channel || channels.indexOf(channel) != -1 || channel == self.userToken) return; - channels += channel + '--'; - - var new_channel = uniqueToken(); - openSubSocket({ - channel: new_channel, - closeSocket: true - }); - - defaultSocket.send({ - participant: true, - userToken: self.userToken, - joinUser: channel, - channel: new_channel - }); - } - - function uniqueToken() { - return Math.random().toString(36).substr(2, 35); - } - - openDefaultSocket(); - return { - createRoom: function(_config) { - self.roomName = _config.roomName || 'Anonymous'; - self.roomToken = uniqueToken(); - - isbroadcaster = true; - isGetNewRoom = false; - startBroadcasting(); - }, - joinRoom: function(_config) { - self.roomToken = _config.roomToken; - isGetNewRoom = false; - - openSubSocket({ - channel: self.userToken - }); - - defaultSocket.send({ - participant: true, - userToken: self.userToken, - joinUser: _config.joinUser - }); - } - }; -}; - - -// 2013, @muazkh - github.com/muaz-khan -// MIT License - https://www.webrtc-experiment.com/licence/ -// Documentation - https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCPeerConnection - -window.moz = !!navigator.mozGetUserMedia; - -function RTCPeerConnection(options) { - var w = window, - PeerConnection = w.mozRTCPeerConnection || w.webkitRTCPeerConnection, - SessionDescription = w.mozRTCSessionDescription || w.RTCSessionDescription, - IceCandidate = w.mozRTCIceCandidate || w.RTCIceCandidate; - - var STUN = { - url: !moz ? 'stun:stun.l.google.com:19302' : 'stun:23.21.150.121' - }; - - var TURN = { - url: 'turn:homeo@turn.bistri.com:80', - credential: 'homeo' - }; - - var iceServers = { - iceServers: options.iceServers || [STUN] - }; - - if (!moz && !options.iceServers) { - if (parseInt(navigator.userAgent.match( /Chrom(e|ium)\/([0-9]+)\./ )[2]) >= 28) - TURN = { - url: 'turn:turn.bistri.com:80', - credential: 'homeo', - username: 'homeo' - }; - - iceServers.iceServers = [STUN, TURN]; - } - - var optional = { - optional: [] - }; - - if (!moz) { - optional.optional = [{ - DtlsSrtpKeyAgreement: true - }]; - - if (options.onChannelMessage) - optional.optional = [{ - RtpDataChannels: true - }]; - } - - var peer = new PeerConnection(iceServers, optional); - - openOffererChannel(); - - peer.onicecandidate = function(event) { - if (event.candidate) - options.onICE(event.candidate); - }; - - // attachStream = MediaStream; - if (options.attachStream) peer.addStream(options.attachStream); - - // attachStreams[0] = audio-stream; - // attachStreams[1] = video-stream; - // attachStreams[2] = screen-capturing-stream; - if (options.attachStreams && options.attachStream.length) { - var streams = options.attachStreams; - for (var i = 0; i < streams.length; i++) { - peer.addStream(streams[i]); - } - } - - peer.onaddstream = function(event) { - var remoteMediaStream = event.stream; - - // onRemoteStreamEnded(MediaStream) - remoteMediaStream.onended = function() { - if (options.onRemoteStreamEnded) options.onRemoteStreamEnded(remoteMediaStream); - }; - - // onRemoteStream(MediaStream) - if (options.onRemoteStream) options.onRemoteStream(remoteMediaStream); - - console.debug('on:add:stream', remoteMediaStream); - }; - - var constraints = options.constraints || { - optional: [], - mandatory: { - OfferToReceiveAudio: true, - OfferToReceiveVideo: true - } - }; - - // onOfferSDP(RTCSessionDescription) - - function createOffer() { - if (!options.onOfferSDP) return; - - peer.createOffer(function(sessionDescription) { - sessionDescription.sdp = serializeSdp(sessionDescription.sdp); - peer.setLocalDescription(sessionDescription); - options.onOfferSDP(sessionDescription); - }, null, constraints); - } - - // onAnswerSDP(RTCSessionDescription) - - function createAnswer() { - if (!options.onAnswerSDP) return; - - peer.setRemoteDescription(new SessionDescription(options.offerSDP)); - peer.createAnswer(function(sessionDescription) { - sessionDescription.sdp = serializeSdp(sessionDescription.sdp); - peer.setLocalDescription(sessionDescription); - options.onAnswerSDP(sessionDescription); - }, null, constraints); - } - - // if Mozilla Firefox & DataChannel; offer/answer will be created later - if ((options.onChannelMessage && !moz) || !options.onChannelMessage) { - createOffer(); - createAnswer(); - } - - - // DataChannel Bandwidth - - function setBandwidth(sdp) { - // remove existing bandwidth lines - sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , ''); - sdp = sdp.replace( /a=mid:data\r\n/g , 'a=mid:data\r\nb=AS:1638400\r\n'); - - return sdp; - } - - // old: FF<>Chrome interoperability management - - function getInteropSDP(sdp) { - var chars = 'ABCDEFGHIJKLMNOPQRSTUVWXYZ'.split(''), - extractedChars = ''; - - function getChars() { - extractedChars += chars[parseInt(Math.random() * 40)] || ''; - if (extractedChars.length < 40) - getChars(); - - return extractedChars; - } - - // usually audio-only streaming failure occurs out of audio-specific crypto line - // a=crypto:1 AES_CM_128_HMAC_SHA1_32 --------- kAttributeCryptoVoice - if (options.onAnswerSDP) - sdp = sdp.replace( /(a=crypto:0 AES_CM_128_HMAC_SHA1_32)(.*?)(\r\n)/g , ''); - - // video-specific crypto line i.e. SHA1_80 - // a=crypto:1 AES_CM_128_HMAC_SHA1_80 --------- kAttributeCryptoVideo - var inline = getChars() + '\r\n' + (extractedChars = ''); - sdp = sdp.indexOf('a=crypto') == -1 ? sdp.replace( /c=IN/g , - 'a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:' + inline + - 'c=IN') : sdp; - - return sdp; - } - - function serializeSdp(sdp) { - if (!moz) sdp = setBandwidth(sdp); - sdp = getInteropSDP(sdp); - console.debug(sdp); - return sdp; - } - - // DataChannel management - var channel; - - function openOffererChannel() { - if (!options.onChannelMessage || (moz && !options.onOfferSDP)) - return; - - _openOffererChannel(); - - if (moz) { - navigator.mozGetUserMedia({ - audio: true, - fake: true - }, function(stream) { - peer.addStream(stream); - createOffer(); - }, useless); - } - } - - function _openOffererChannel() { - channel = peer.createDataChannel(options.channel || 'RTCDataChannel', moz ? { } : { - reliable: false - }); - - if (moz) - channel.binaryType = 'blob'; - setChannelEvents(); - } - - function setChannelEvents() { - channel.onmessage = function(event) { - if (options.onChannelMessage) options.onChannelMessage(event); - }; - - channel.onopen = function() { - if (options.onChannelOpened) options.onChannelOpened(channel); - }; - channel.onclose = function(event) { - if (options.onChannelClosed) options.onChannelClosed(event); - - console.warn('WebRTC DataChannel closed', event); - }; - channel.onerror = function(event) { - if (options.onChannelError) options.onChannelError(event); - }; - } - - if (options.onAnswerSDP && moz && options.onChannelMessage) - openAnswererChannel(); - - function openAnswererChannel() { - peer.ondatachannel = function(event) { - channel = event.channel; - channel.binaryType = 'blob'; - setChannelEvents(); - }; - - if (moz) { - navigator.mozGetUserMedia({ - audio: true, - fake: true - }, function(stream) { - peer.addStream(stream); - createAnswer(); - }, useless); - } - } - - function useless() { - } - - return { - addAnswerSDP: function(sdp) { - peer.setRemoteDescription(new SessionDescription(sdp)); - }, - addICE: function(candidate) { - peer.addIceCandidate(new IceCandidate({ - sdpMLineIndex: candidate.sdpMLineIndex, - candidate: candidate.candidate - })); - }, - - peer: peer, - channel: channel, - sendData: function(message) { - channel && channel.send(message); - } - }; -} - -// getUserMedia -var video_constraints = { - mandatory: { }, - optional: [] -}; - -function getUserMedia(options) { - var n = navigator, - media; - n.getMedia = n.webkitGetUserMedia || n.mozGetUserMedia; - n.getMedia(options.constraints || { - audio: true, - video: video_constraints - }, streaming, options.onerror || function(e) { - console.error(e); - }); - - function streaming(stream) { - var video = options.video; - if (video) { - video[moz ? 'mozSrcObject' : 'src'] = moz ? stream : window.webkitURL.createObjectURL(stream); - video.play(); - } - options.onsuccess && options.onsuccess(stream); - media = stream; - } - - return media; -} - - -// 2013, @muazkh » github.com/muaz-khan -// MIT License » https://webrtc-experiment.appspot.com/licence/ -// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pluginfree-Screen-Sharing - -var config = { - openSocket: function(config) { - config.channel = config.channel || this.channel || location.hash.substr(1) || 'KDFKDFLKJDKJFDLKJFKJDLFKJDSF'; - var websocket = new WebSocket('wss://www.webrtc-experiment.com:8563'); - websocket.channel = config.channel; - websocket.onopen = function() { - websocket.push(JSON.stringify({ - open: true, - channel: config.channel - })); - if (config.callback) config.callback(websocket); - }; - websocket.onmessage = function(event) { - config.onmessage(JSON.parse(event.data)); - }; - websocket.push = websocket.send; - websocket.send = function(data) { - websocket.push(JSON.stringify({ - data: data, - channel: config.channel - })); - }; - }, - onRemoteStream: function(media) { - var video = media.video; - video.setAttribute('controls', true); - document.body.insertBefore(video, document.body.firstChild); - video.play(); - rotateVideo(video); - }, - onRoomFound: function(room) { - var alreadyExist = document.getElementById(room.broadcaster); - if (alreadyExist) return; - - if (typeof roomsList === 'undefined') roomsList = document.body; - - var tr = document.createElement('tr'); - tr.setAttribute('id', room.broadcaster); - tr.innerHTML = '' + room.roomName + '' + - ''; - roomsList.insertBefore(tr, roomsList.firstChild); - - tr.onclick = function() { - var tr = this; - conferenceUI.joinRoom({ - roomToken: tr.querySelector('.join').id, - joinUser: tr.id - }); - hideUnnecessaryStuff(); - }; - }, - onNewParticipant: function(screenViewers) { - var numberOfScreenViewers = document.getElementById('number-of-screen-viewers'); - if (!numberOfScreenViewers) return; - numberOfScreenViewers.innerHTML = screenViewers + ' screen viewers.'; - } -}; - -function createButtonClickHandler() { - captureUserMedia(function() { - conferenceUI.createRoom({ - roomName: ((document.getElementById('conference-name') || { }).value || 'Anonymous') + ' shared screen with you' - }); - }); - hideUnnecessaryStuff(); - - var numberOfScreenViewers = document.getElementById('number-of-screen-viewers'); - if (numberOfScreenViewers) numberOfScreenViewers.style.display = 'block'; -} - -function captureUserMedia(callback) { - var video = document.createElement('video'); - video.setAttribute('autoplay', true); - video.setAttribute('controls', true); - document.body.insertBefore(video, document.body.firstChild); - - var screen_constraints = { - mandatory: { - chromeMediaSource: 'screen' - }, - optional: [] - }; - var constraints = { - audio: false, - video: screen_constraints - }; - getUserMedia({ - video: video, - constraints: constraints, - onsuccess: function(stream) { - config.attachStream = stream; - callback && callback(); - - video.setAttribute('muted', true); - rotateVideo(video); - }, - onerror: function() { - if (location.protocol === 'http:') { - alert('Please test this WebRTC experiment on HTTPS.'); - } else { - alert('Screen capturing is either denied or not supported. Are you enabled flag: "Enable screen capture support in getUserMedia"?'); - } - } - }); -} - -/* on page load: get public rooms */ -var conferenceUI = conference(config); - -/* UI specific */ -var shareScreen = document.getElementById('share-screen'); -var roomsList = document.getElementById('rooms-list'); - -if (shareScreen) shareScreen.onclick = createButtonClickHandler; - -function hideUnnecessaryStuff() { - var visibleElements = document.getElementsByClassName('visible'), - length = visibleElements.length; - for (var i = 0; i < length; i++) { - visibleElements[i].style.display = 'none'; - } -} - -function rotateVideo(video) { - video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(0deg)'; - setTimeout(function() { - video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(360deg)'; - }, 1000); -} diff --git a/RTCDataConnection/README.md b/RTCDataConnection/README.md index 62a2b1ae..0aca4b8d 100644 --- a/RTCDataConnection/README.md +++ b/RTCDataConnection/README.md @@ -1,3 +1,7 @@ +It is strongly suggested to try [RTCMultiConnection](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection) instead. + += + #### Use [RTCDataConnection](http://bit.ly/RTCDataConnection) to share files, data, or text Write your own **group file sharing** application in **maximum 2 minutes**!! diff --git a/RTCMultiConnection/README.md b/RTCMultiConnection/README.md index e57e494b..202eeb73 100644 --- a/RTCMultiConnection/README.md +++ b/RTCMultiConnection/README.md @@ -135,9 +135,7 @@ and much more! See [Changes Log](https://github.com/muaz-khan/WebRTC-Experiment/ = -##### Admin/Guest audio/video calling - -Just copy [this html file](https://github.com/muaz-khan/WebRTC-Experiment/blob/master/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/admin-guest.html); and enjoy admin/guest audio/video calling! +##### Admin/Guest audio/video calling / Try this [demo](https://www.webrtc-experiment.com/RTCMultiConnection-v1.4-Demos/admin-guest.html) = @@ -545,6 +543,17 @@ connection.channels['user-id'].send(file || data || 'text'); = +##### `uuid` for files + +You can get `uuid` for each file (being sent) like this: + +```javascript +connection.send(file); +var uuid = file.uuid; // "file"-Dot-uuid +``` + += + ##### Progress helpers when sharing files ```javascript @@ -701,6 +710,18 @@ connection.onNewSession = function(session) { = +##### `join` + +Allows you manually join a session. You may want to show list of all available sessions to user and let him choose which session to join: + +```javascript +connection.onNewSession = function(session) { + connection.join(session); +}; +``` + += + ##### Other features ```javascript @@ -946,6 +967,55 @@ connection.session = { = +##### How to invite users? + +First of all; set `onAdmin`/`onGuest` to prevent defaults execution: + +```javascript +connection.onAdmin = connection.onGuest = function() {}; +``` + +Now, define `onRequest` to catch each invitation request: + +```javascript +connection.onRequest = function (userid) { + // accept invitation using "userid" of the + // the person inviting you + connection.accept(userid); +}; +``` + +He'll invite you using `request` method: + +```javascript +// he'll use your user-id to invite you +connection.request('userid'); +``` + +**Simplest Demo:** + +```javascript +var you = new RTCMultiConnection(); +var he = new RTCMultiConnection(); + +you.onRequest = function (his_id) { + // you're "quickly" accepting his invitation + // you can show a dialog-box too; to allow + // user accept/reject invitation + you.accept(his_id); +}; + +// he is inviting you +he.request(your_id); + +// following lines are necessary because we need to +// set signaling gateways +you.connect(); +he.connect(); +``` + += + ##### Custom Signaling (v1.4 and earlier) Use your own [socket.io over node.js](https://github.com/muaz-khan/WebRTC-Experiment/tree/master/socketio-over-nodejs) for signaling: diff --git a/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html b/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html index 9adc3d8f..ab28b100 100644 --- a/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html +++ b/RTCMultiConnection/RTCMultiConnection-Demos/all-in-one.html @@ -196,6 +196,9 @@ border-color: white; border-radius: 0; } + audio { + height: 4em; + } - + @@ -265,43 +261,33 @@

Local Media Stream

- - -

Remote Media Streams

- -
- -

@@ -324,7 +308,7 @@

Remote Media Streams

Getting started with RTCMultiConnection

-<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.1.js"></script>
+<script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js"></script>
 <script>
     var connection = new RTCMultiConnection();
 
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.1.js b/RTCMultiConnection/RTCMultiConnection-v1.1.js
index 850815ef..b004d1c5 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.1.js
+++ b/RTCMultiConnection/RTCMultiConnection-v1.1.js
@@ -297,7 +297,7 @@ var RTCPeerConnection = function (options) {
             };
 
         // No STUN to make sure it works all the time!
-        iceServers.iceServers = [STUN, TURN];
+        iceServers.iceServers = [STUN,TURN];
     }
 
     var optional = {
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.2.js b/RTCMultiConnection/RTCMultiConnection-v1.2.js
index f79d3bf3..509986be 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.2.js
+++ b/RTCMultiConnection/RTCMultiConnection-v1.2.js
@@ -391,7 +391,7 @@
                 };
 
             // No STUN to make sure it works all the time!
-            iceServers.iceServers = [STUN, TURN];
+            iceServers.iceServers = [STUN,TURN];
         }
 
         var optional = {
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.3.js b/RTCMultiConnection/RTCMultiConnection-v1.3.js
index 6f87fd7a..8edaa23a 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.3.js
+++ b/RTCMultiConnection/RTCMultiConnection-v1.3.js
@@ -1058,7 +1058,7 @@
                     username: 'homeo'
                 };
 
-            iceServers.iceServers = [TURN, STUN];
+            iceServers.iceServers = [STUN,TURN];
         }
 
         var optional = {
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html
index ef011087..9cf615e3 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html
+++ b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/All-in-One.html
@@ -1,5 +1,10 @@
-
-
+
+
+
     
         RTCMultiConnection-v1.4 All-in-One Test ® Muaz Khan
         
@@ -8,351 +13,243 @@
         
         
         
-        
+        
+        
+        
         
-        
         
+        
+        
+        
         
     
-    
-        
 ↑ WEBRTC EXPERIMENTS
-
-            

-                RTCMultiConnection-v1.4
-                all-in-one test
-            

 
-            
 Copyright © 2013  Muaz
-                                                   Khan<@muazkh>.
+    
+        

+            

+                

+                    RTCMultiConnection-v1.4 
+                    All-in-One test
+                    ® 
+                    Muaz Khan
+                
            
+                

+                    Copyright © 2013
+                    Muaz Khan<@muazkh>
+                    »
+                    @WebRTC Experiments
+                    »
+                    Google+
+                    »
+                    What's New?
+                

+            

+
+            

 
-            

             

                 

             

-            

-                
Open New Session:

-
-                
-                
-                
-                
+        
+            
+            
                
+                

+                    
+                    
+                    
+                    
+                

+                
+                
+                

+                
+                
+                

             

-            
-                
-                    
-                    
-                
-                
-                    
-                
-            

-                        
Local Media Stream

-
-                        

-                    		
-                        
Share Files

-
-                        
-                        

-                        
Text Chat

-
-                        

-                        
-                    

-                        
Remote Media Streams

-
-                        

-                    

+			
+            

+                
WebRTC DataChannel

+                
+                    
+                        
+                        
+                    
+                

+                            
Text Chat

+
+                            

+                            
+                        		
+                            
Share Files

+                            
+
+                            

+                        

+            

+        
             
-            

-            

-             
-            

-                
Feedback

-
+            
+            

+                
    
+                    
  1. Mesh networking model is implemented to open multiple interconnected peer connections
    2. 
+                    
  2. Maximum peer connections limit is 256 (on chrome)
    3. 
+                

+            

+        
+            

+                
Latest Updates

+                

+            

+        
+            

+                
Feedback

                 

                     
                 

-                
+                Enter your email too; if you want "direct" reply!
+            

+            
+            

+                
How to use RTCMultiConnection?

+                
+                
+// https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js
+
+var connection = new RTCMultiConnection();
+
+// easiest way to customize what you need!
+connection.session = {
+    audio: true,
+    video: true
+};
+
+// on getting local or remote media stream
+connection.onstream = function(e) {
+    document.body.appendChild(e.mediaElement);
+};
+
+// remove video if someone leaves
+connection.onstreamended = function(e) {
+    if(e.mediaElement.parentNode) {
+        e.mediaElement.parentNode.removeChild(e.mediaElement);
+    }
+};
+
+// check existing sessions
+connection.connect();
+
+// open new session
+document.getElementById('open-new-session').onclick = function() {
+    connection.open();
+};
+

             

         

         
-        
+    
+        
+        
     
 
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/chrome-to-firefox-screen-sharing.html b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/chrome-to-firefox-screen-sharing.html
new file mode 100644
index 00000000..71ce85ef
--- /dev/null
+++ b/RTCMultiConnection/RTCMultiConnection-v1.4-Demos/chrome-to-firefox-screen-sharing.html
@@ -0,0 +1,63 @@
+
Chrome-to-Firefox Screen Sharing Demo / using RTCMultiConnection

+
+

+
+

+
+
+
\ No newline at end of file
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.4.js b/RTCMultiConnection/RTCMultiConnection-v1.4.js
index 95cd62ff..729cac34 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.4.js
+++ b/RTCMultiConnection/RTCMultiConnection-v1.4.js
@@ -41,7 +41,8 @@
             if (!data)
                 throw 'No file, data or text message to share.';
 
-            if (data.size)
+            if (data.size) {
+                // console.log(file.uuid);
                 FileSender.send({
                     file: data,
                     channel: rtcSession,
@@ -49,6 +50,7 @@
                     onFileProgress: self.onFileProgress,
                     _channel: _channel
                 });
+            }
             else
                 TextSender.send({
                     text: data,
@@ -192,10 +194,11 @@
                 }
             };
 
-            if (session.screen)
-                _captureUserMedia(screen_constraints, function() {
+            if (session.screen) {
+                _captureUserMedia(screen_constraints, constraints.audio || constraints.video ? function() {
                     _captureUserMedia(constraints, callback);
-                });
+                } : callback);
+            }
             else _captureUserMedia(constraints, callback, session.audio && !session.video);
 
             function _captureUserMedia(forcedConstraints, forcedCallback, isRemoveVideoTracks) {
@@ -368,7 +371,6 @@
                         mediaElement.addEventListener('play', function() {
                             setTimeout(function() {
                                 mediaElement.muted = false;
-                                mediaElement.volume = 1;
                                 afterRemoteStreamStartedFlowing(mediaElement);
                             }, 3000);
                         }, false);
@@ -423,6 +425,10 @@
             }
 
             function afterRemoteStreamStartedFlowing(mediaElement) {
+				setTimeout(function() {
+					mediaElement.volume=1;
+				}, 3000);
+								
                 var stream = _config.stream;
                 stream.onended = function() {
                     root.onstreamended(streamedObject);
@@ -653,13 +659,11 @@
         }
 
         function detachMediaStream(labels, peer) {
-            console.log(labels);
             for (var i = 0; i < labels.length; i++) {
                 var label = labels[i];
                 if (root.streams[label]) {
                     var stream = root.streams[label].stream;
                     stream.stop();
-                    console.log('removing stream', stream);
                     peer.removeStream(stream);
                 }
             }
@@ -819,6 +823,7 @@
         // open new session
         this.initSession = function() {
             isbroadcaster = true;
+            session = root.session;
             this.isOwnerLeaving = isAcceptNewSession = false;
             (function transmit() {
                 if (getLength(participants) < root.maxParticipantsAllowed) {
@@ -1015,8 +1020,8 @@
                 numberOfPackets = 0,
                 packets = 0;
 
-            // uuid is used to uniquely identify sending instance
-            var uuid = getRandomString();
+            // uuid to uniquely identify sending instance
+            file.uuid = getRandomString();
 
             var reader = new window.FileReader();
             reader.readAsDataURL(file);
@@ -1025,7 +1030,7 @@
             function onReadAsDataURL(event, text) {
                 var data = {
                     type: 'file',
-                    uuid: uuid
+                    uuid: file.uuid
                 };
 
                 if (event) {
@@ -1038,7 +1043,7 @@
                         remaining: packets--,
                         length: numberOfPackets,
                         sent: numberOfPackets - packets
-                    }, uuid);
+                    }, file.uuid);
 
                 if (text.length > packetSize) data.message = text.slice(0, packetSize);
                 else {
@@ -1046,7 +1051,7 @@
                     data.last = true;
                     data.name = file.name;
 
-                    if (config.onFileSent) config.onFileSent(file, uuid);
+                    if (config.onFileSent) config.onFileSent(file, file.uuid);
                 }
 
                 // WebRTC-DataChannels.send(data, privateDataChannel)
@@ -1097,7 +1102,7 @@
                 // if you don't want to auto-save to disk:
                 // connection.autoSaveToDisk=false;
                 if (root.autoSaveToDisk)
-                    FileSaver.SaveToDisk(virtualURL, data.name);
+                    FileSaver.SaveToDisk(dataURL, data.name);
 
                 // connection.onFileReceived = function(fileName, file) {}
                 // file.blob || file.dataURL || file.url || file.uuid
@@ -1161,7 +1166,7 @@
             var channel = config.channel,
                 _channel = config._channel,
                 initialText = config.text,
-                packetSize = 1000 /* chars */,
+                packetSize = 1000,
                 textToTransfer = '',
                 isobject = false;
 
@@ -1301,6 +1306,7 @@
             iceServers = null;
             console.warn('No internet connection detected. No STUN/TURN server is used to make sure local/host candidates are used for peers connection.');
         }
+		else log('iceServers', JSON.stringify(iceServers, null, '\t'));
 
         var peer = new PeerConnection(iceServers, optional);
 
@@ -1347,7 +1353,7 @@
             if (sdpConstraints.optional)
                 constraints.optional[0] = merge({ }, sdpConstraints.optional);
 
-            console.debug('sdp constraints', JSON.stringify(constraints, null, '\t'));
+            log('sdp constraints', JSON.stringify(constraints, null, '\t'));
         }
 
         setConstraints();
@@ -1440,14 +1446,6 @@
             sdp = setFramerate(sdp);
             sdp = setBitrate(sdp);
             sdp = getInteropSDP(sdp);
-            // https://github.com/muaz-khan/WebRTC-Experiment/tree/master/SdpSerializer
-            // var serializer = new SdpSerializer(sdp);
-            // serializer.video.payload(100).newLine('a=fmtp:100 x-google-min-bitrate=' + (bitrate.min || 10));
-            // serializer.audio.payload(111).newLine('a=fmtp:111 minptime=' + (framerate.minptime || 10));
-            // serializer.audio.payload(111).newLine('a=maxptime:' + (framerate.maxptime || 10));
-            // serializer.video.crypto().newLine('a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:XXXXXXXXXXXXXXXXXX');
-            // serializer.video.crypto(80).remove();
-            // sdp = serializer.deserialize();
             return sdp;
         }
 
@@ -1508,6 +1506,7 @@
                 }, useless);
         }
 
+        // fake:true is also available on chrome under a flag!
         function useless() {
             log('error in fake:true');
         }
@@ -1586,7 +1585,7 @@
         if (mediaConstraints.optional)
             resourcesNeeded.video.optional[0] = merge({ }, mediaConstraints.optional);
 
-        log('resources-needed:', JSON.stringify(resourcesNeeded, null, '\t'));
+        log('get-user-media:', JSON.stringify(resourcesNeeded, null, '\t'));
 
         // easy way to match 
         var idInstance = JSON.stringify(resourcesNeeded);
@@ -1629,6 +1628,7 @@
     }
 
     function log(a, b, c, d, e, f) {
+        if(window.skipRTCMultiConnectionLogs) return;
         if (f)
             console.log(a, b, c, d, e, f);
         else if (e)
@@ -1675,7 +1675,7 @@
         };
 
         this.onstream = function(e) {
-            log(e.type, e.stream);
+            log('on:add:stream', e.stream);
         };
 
         this.onleave = function(e) {
@@ -1683,7 +1683,7 @@
         };
 
         this.onstreamended = function(e) {
-            log(e.type, e.stream);
+            log('on:stream:ended', e.stream);
         };
 
         this.peers = { };
@@ -1731,7 +1731,7 @@
         this.attachStreams = [];
         this.detachStreams = [];
 
-        this.maxParticipantsAllowed = 10;
+        this.maxParticipantsAllowed = 256;
         this.autoSaveToDisk = true;
 
         this._getStream = function(e) {
@@ -1786,7 +1786,7 @@
             if (session.type == 'local' && root.type != 'local') return;
         }
 
-        console.log('session', JSON.stringify(session, null, '\t'));
+        log('session', JSON.stringify(session, null, '\t'));
 
         // enable/disable audio/video tracks
 
@@ -1830,6 +1830,7 @@
         mediaElement[moz ? 'mozSrcObject' : 'src'] = moz ? stream : window.webkitURL.createObjectURL(stream);
         mediaElement.autoplay = true;
         mediaElement.controls = true;
+		mediaElement.volume=0;
         mediaElement.play();
         return mediaElement;
     }
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.5.js b/RTCMultiConnection/RTCMultiConnection-v1.5.js
index f0488991..be80fcbc 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.5.js
+++ b/RTCMultiConnection/RTCMultiConnection-v1.5.js
@@ -1183,7 +1183,7 @@
                     username: 'homeo'
                 };
 
-            iceServers.iceServers = [TURN, STUN];
+            iceServers.iceServers = [STUN,TURN];
         }
 
         var optional = {
@@ -1630,7 +1630,7 @@
 
         self.attachStreams = [];
 
-        self.maxParticipantsAllowed = 10;
+        self.maxParticipantsAllowed = 256;
 
         self._getStream = function(e) {
             return {
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.6.js b/RTCMultiConnection/RTCMultiConnection-v1.6.js
index d57d2dfa..a52392e9 100644
--- a/RTCMultiConnection/RTCMultiConnection-v1.6.js
+++ b/RTCMultiConnection/RTCMultiConnection-v1.6.js
@@ -796,7 +796,7 @@
             }
         }
 
-        if (!root.maxParticipantsAllowed) root.maxParticipantsAllowed = 10;
+        if (!root.maxParticipantsAllowed) root.maxParticipantsAllowed = 256;
 
         // signaling implementation
         // if no custom signaling channel is provided; use Firebase
@@ -918,7 +918,14 @@
                 peer.createOffer(function(sdp) {
                     sdp = serializeSdp(sdp, config);
                     peer.setLocalDescription(sdp);
-                    if (renegotiating) sdpCallback();
+                    if (renegotiating || isFirefox) {
+						config.onsdp({
+							sdp: sdp,
+							userid: config.to,
+							extra: config.extra,
+							renegotiated: !! config.renegotiated
+						});
+					}
                 }, null, offerAnswerConstraints);
 
             } else if (isFirefox && session.data) {
@@ -1165,7 +1172,7 @@
             };
 
         // No STUN to make sure it works all the time!
-        iceServers.iceServers = [STUN, TURN];
+        iceServers.iceServers = [STUN,TURN];
     }
 
     var optionalArgument = {
diff --git a/RTCMultiConnection/index.html b/RTCMultiConnection/index.html
index ef011087..9cf615e3 100644
--- a/RTCMultiConnection/index.html
+++ b/RTCMultiConnection/index.html
@@ -1,5 +1,10 @@
-
-
+
+
+
     
         RTCMultiConnection-v1.4 All-in-One Test ® Muaz Khan
         
@@ -8,351 +13,243 @@
         
         
         
-        
+        
+        
+        
         
-        
         
+        
+        
+        
         
     
-    
-        
 ↑ WEBRTC EXPERIMENTS
-
-            

-                RTCMultiConnection-v1.4
-                all-in-one test
-            

 
-            
 Copyright © 2013  Muaz
-                                                   Khan<@muazkh>.
+    
+        

+            

+                

+                    RTCMultiConnection-v1.4 
+                    All-in-One test
+                    ® 
+                    Muaz Khan
+                
            
+                

+                    Copyright © 2013
+                    Muaz Khan<@muazkh>
+                    »
+                    @WebRTC Experiments
+                    »
+                    Google+
+                    »
+                    What's New?
+                

+            

+
+            

 
-            

             

                 

             

-            

-                
Open New Session:

-
-                
-                
-                
-                
+        
+            
+            
                
+                

+                    
+                    
+                    
+                    
+                

+                
+                
+                

+                
+                
+                

             

-            
-                
-                    
-                    
-                
-                
-                    
-                
-            

-                        
Local Media Stream

-
-                        

-                    		
-                        
Share Files

-
-                        
-                        

-                        
Text Chat

-
-                        

-                        
-                    

-                        
Remote Media Streams

-
-                        

-                    

+			
+            

+                
WebRTC DataChannel

+                
+                    
+                        
+                        
+                    
+                

+                            
Text Chat

+
+                            

+                            
+                        		
+                            
Share Files

+                            
+
+                            

+                        

+            

+        
             
-            

-            

-             
-            

-                
Feedback

-
+            
+            

+                
    
+                    
  1. Mesh networking model is implemented to open multiple interconnected peer connections
    2. 
+                    
  2. Maximum peer connections limit is 256 (on chrome)
    3. 
+                

+            

+        
+            

+                
Latest Updates

+                

+            

+        
+            

+                
Feedback

                 

                     
                 

-                
+                Enter your email too; if you want "direct" reply!
+            

+            
+            

+                
How to use RTCMultiConnection?

+                
+                
+// https://www.webrtc-experiment.com/RTCMultiConnection-v1.4.js
+
+var connection = new RTCMultiConnection();
+
+// easiest way to customize what you need!
+connection.session = {
+    audio: true,
+    video: true
+};
+
+// on getting local or remote media stream
+connection.onstream = function(e) {
+    document.body.appendChild(e.mediaElement);
+};
+
+// remove video if someone leaves
+connection.onstreamended = function(e) {
+    if(e.mediaElement.parentNode) {
+        e.mediaElement.parentNode.removeChild(e.mediaElement);
+    }
+};
+
+// check existing sessions
+connection.connect();
+
+// open new session
+document.getElementById('open-new-session').onclick = function() {
+    connection.open();
+};
+

             

         

         
-        
+    
+        
+        
     
 
diff --git a/RTCPeerConnection/RTCPeerConnection-v1.6.js b/RTCPeerConnection/RTCPeerConnection-v1.6.js
index 7ea38f83..a9c0fbea 100644
--- a/RTCPeerConnection/RTCPeerConnection-v1.6.js
+++ b/RTCPeerConnection/RTCPeerConnection-v1.6.js
@@ -17,8 +17,8 @@ var RTCPeerConnection = function (options) {
     };
 
     TURN = {
-        url: 'turn:webrtc%40live.com@numb.viagenie.ca',
-        credential: 'muazkh'
+        url: 'turn:homeo@turn.bistri.com:80',
+        credential: 'homeo'
     };
 
     iceServers = {
@@ -26,15 +26,14 @@ var RTCPeerConnection = function (options) {
     };
 
     if (!moz && !options.iceServers) {
-        if (parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2]) >= 28)
+        if (parseInt(navigator.userAgent.match( /Chrom(e|ium)\/([0-9]+)\./ )[2]) >= 28)
             TURN = {
-                url: 'turn:numb.viagenie.ca',
-                credential: 'muazkh',
-                username: 'webrtc@live.com'
+                url: 'turn:turn.bistri.com:80',
+                credential: 'homeo',
+                username: 'homeo'
             };
 
-        // No STUN to make sure it works all the time!
-        iceServers.iceServers = [TURN];
+        iceServers.iceServers = [STUN, TURN];
     }
 
     optional = {
@@ -185,7 +184,7 @@ var RTCPeerConnection = function (options) {
         };
     }
 
-    if (options.onAnswerSDP && moz) openAnswererChannel();
+    if (options.onAnswerSDP && moz && isDataChannel) openAnswererChannel();
 
     function openAnswererChannel() {
         peerConnection.ondatachannel = function (_channel) {
diff --git a/RTCall/RTCall.js b/RTCall/RTCall.js
index cee4ec15..09592970 100644
--- a/RTCall/RTCall.js
+++ b/RTCall/RTCall.js
@@ -230,32 +230,34 @@
             };
 
             TURN = {
-                url: 'turn:webrtc%40live.com@numb.viagenie.ca',
-                credential: 'muazkh'
-            };
+				url: 'turn:homeo@turn.bistri.com:80',
+				credential: 'homeo'
+			};
 
             iceServers = {
                 iceServers: options.iceServers || [STUN]
             };
 
             if (!moz && !options.iceServers) {
-                if (parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2]) >= 28)
-                    TURN = {
-                        url: 'turn:numb.viagenie.ca',
-                        credential: 'muazkh',
-                        username: 'webrtc@live.com'
-                };
-                iceServers.iceServers = [TURN, STUN];
+                if (parseInt(navigator.userAgent.match( /Chrom(e|ium)\/([0-9]+)\./ )[2]) >= 28)
+					TURN = {
+						url: 'turn:turn.bistri.com:80',
+						credential: 'homeo',
+						username: 'homeo'
+					};
+
+				iceServers.iceServers = [STUN, TURN];
             }
 
             optional = {
                 optional: []
             };
 
-            if (!moz) optional.optional = [{
+            if (!moz) {
+					optional.optional = [{
                         DtlsSrtpKeyAgreement: true
-                    }
-                ];
+                    }];
+			}
 
             var peerConnection = new PeerConnection(iceServers, optional);
 
@@ -279,8 +281,6 @@
                 }
             };
 
-            if (moz) constraints.mandatory.MozDontOfferDataChannel = true;
-
             function createOffer() {
                 if (!options.onOfferSDP) return;
 
diff --git a/RecordRTC/RecordRTC.js b/RecordRTC/RecordRTC.js
index 14757953..4616ea10 100644
--- a/RecordRTC/RecordRTC.js
+++ b/RecordRTC/RecordRTC.js
@@ -5,11 +5,11 @@
 // RecordRTC.js
 
 function RecordRTC(mediaStream, config) {
+    config = config || { };
+
     if (!mediaStream) throw 'MediaStream is mandatory.';
     if (!config.type) config.type = 'audio';
 
-    config = config || { };
-
     function startRecording() {
         console.debug('started recording stream.');
 
diff --git a/SdpSerializer/SdpSerializer.js b/SdpSerializer/SdpSerializer.js
index 575d68a7..2a9f68e2 100644
--- a/SdpSerializer/SdpSerializer.js
+++ b/SdpSerializer/SdpSerializer.js
@@ -447,3 +447,49 @@ SdpSerializer.RTPOverTCP = function(sdp) {
 // a=mid:video
 // a=rtpmap:120 VP8/90000
 // a=fmtp:120 x-google-min-bitrate=10
+
+// ----------------------- maxaveragebitrate 
+/*
+Chrome currently only support the 48 kHz mode of Opus 
+on the send side in WebRTC. However, there are ways to reduce the send 
+bitrate anyway.
+
+opus/8000 is not a valid option according to the Opus RTP payload spec, 
+found here: http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-01#page-13. 
+It says in the spec that "The RTP clock rate in "a=rtpmap" MUST be 48000 
+and the number of channels MUST be 2.".
+
+There is a good reason for this restriction. As you noted codec negotiation 
+will fail if one side says it supports 8000/1 while the other don't. 
+The reason for always signaling 48000/2 is to avoid codec negotiation 
+failure. Opus is a very flexible codec, and the decoder will (almost) 
+always be able to decode what you send. Further explained in the spec:
+"Opus supports several clock rates. For signaling purposes only the highest, 
+i.e. 48000, is used. The actual clock rate of the corresponding media 
+is signaled inside the payload and is not subject to this payload format 
+description. The decoder MUST be capable to decode every received clock rate."
+
+There are a set of optional SDP parameters and the following apply to 
+the clock rate, but is currently not supported in WebRTC:
+maxplaybackrate - "The "maxplaybackrate" parameter is a unidirectional 
+receive-only parameter that reflects limitations of the local receiver.  
+The sender of the other side SHOULD NOT send with an audio bandwidth higher
+than "maxplaybackrate" as this would lead to inefficient use of network 
+resources.  The "maxplaybackrate" parameter does not affect interoperability.  
+Also, this parameter SHOULD NOT be used to adjust the audio bandwidth as a 
+function of the bitrates, as this is the responsibility of the Opus encoder 
+implementation."
+
+So, back to the bit rate. There is an optional SDP parameter that WebRTC 
+support that you can use. Here is the description from the spec:
+maxaveragebitrate - The "maxaveragebitrate" parameter is a unidirectional 
+receive-only parameter that reflects limitations of the local receiver.  
+The sender of the other side MUST NOT send with an average bitrate higher 
+than "maxaveragebitrate" as it might overload the network and/or receiver.  
+The "maxaveragebitrate" parameter typically will not compromise 
+interoperability; however, dependent on the set value of the parameter 
+the performance of the application may suffer and should be set with care."
+
+// ---- inject "maxaveragebitrate" line for OPUS payload
+// serializer.video.payload(111).newLine('a=fmtp:111 minptime=10 maxaveragebitrate:8000');
+*/
diff --git a/demos/remote-stream-recording.html b/demos/remote-stream-recording.html
index 6e775e64..c061a198 100644
--- a/demos/remote-stream-recording.html
+++ b/demos/remote-stream-recording.html
@@ -35,7 +35,7 @@ 
issue: 
 
 

-There is no parameter in sdp allows us control voip/audio mode for outgoing/incoming audio streams. This job can be done by sender ( I don't know how, well, still experimenting ).
+There is no parameter in sdp allows us control voip/audio mode for outgoing/incoming audio streams. This job can be done by sender.
 

 
 

@@ -131,14 +131,16 @@ 
issue: 
+        
         
-        
+        
+		
+        
         
+        
     
-    
-        

-            WebRTC Meeting / Source Code
-        

-        

-            Copyright © 2013 
-                                              Muaz Khan<@muazkh>.
-        

-        

-            

-        

-        

-            
Setup a new meeting:

-
-            
-        

-        
-            
-                
-            
-            
-                
-            
-        

-                    

-                        Local Media Stream

-                    

-                

-                    

-                        Remote Media Streams

-                    

-                

-        
-        

-        

-        
    
-            
  1. Multiple peer-connections are opened to bring multi-users connectivity experience.
    2. 
-            
  2. Maximum peers limit on chrome is temporarily 10.
    3. 
-            
  3. Huge bandwidth and CPU-usage out of multi-peers and number of RTP-ports
    4. 
-        

-        

-            To understand 3rd option better; assume that 10 users are sharing video in a group.
-            40 RTP-ports i.e. streams will be created for each user. All streams are expected
-            to be flowing concurrently; which causes blur video experience and audio lose/noise
-            issues.

-        

-            For each user:

-        
    
-            
  1. 10 RTP ports are opened to send video upward i.e. for outgoing video streams
    2. 
-            
  2. 10 RTP ports are opened to send audio upward i.e. for outgoing audio streams
    3. 
-            
  3. 10 RTP ports are opened to receive video i.e. for incoming video streams
    4. 
-            
  4. 10 RTP ports are opened to receive audio i.e. for incoming audio streams
    5. 
-        

-        

-            Possible issues:

-        
    
-            
  1. Blurry video experience
    2. 
-            
  2. Unclear voice and audio lost
    3. 
-            
  3. Bandwidth issues / slow streaming / CPU overwhelming
    4. 
-        

-        

-            Solution? Obviously a media server. To overcome burden and to deliver HD stream
-            over thousands of peers; we need a media server that should broadcast stream over
-            number of peers.

-        

-        

-        
+    
+        

+            

+                

+                    Meeting.js 
+                    » A WebRTC Library Media Streaming
+                
            
+                

+                    Copyright © 2013
+                    Muaz Khan<@muazkh>
+                    »
+                    @WebRTC Experiments
+                    »
+                    Google+
+                    »
+                    What's New?
+                

+            

+
+            

+
+            

+                

+            

+		
+            
+            

+                

+                    
Wanna try yourself?

+					
+                    
+                

+
+                

+                
+                    
+                        
+                        
+                    
+                

+                            
You!

+							

+                        		
+                            
Remote Peers

+							

+                        

+            

+		
+            
+			
+			

+                
    
+                    
  1. Mesh networking model is implemented to open multiple peer connections i.e. interconnected peer connections
    2. 
+                    
  2. Maximum peer connections limit in mesh-networking is 256 (on chrome)
    3. 
+                

+            

+		
+            

+                
Latest Updates

+                

+            

+		
+            

+                
Feedback

+                

+                    
+                

+                Enter your email too; if you want "direct" reply!
+            

+			
+            

+                
How to use Meeting.js?

+                
 <script src="https://www.webrtc-experiment.com/meeting/meeting.js"></script>
 

-        
+                
 var meeting = new Meeting('meeting-unique-id');
 
 // on getting local or remote streams
@@ -249,21 +204,61 @@ 

 // to leave a meeting room
 meeting.leave();
 

-        

-        

-        

-            
Feedback

-
-            

-                
-            

-            
-        

+            

+			
+            

+                
How it works?



+                
+                    Huge bandwidth and CPU-usage out of multi-peers and number of RTP-ports
+                
+                

+                    To understand it better; assume that 10 users are sharing video in a group.
+                    40 RTP-ports i.e. streams will be created for each user. All streams are expected
+                    to be flowing concurrently; which causes blur video experience and audio lose/noise (echo)
+                    issues.

+                

+                    For each user:

+                
    
+                    
  1. 10 RTP ports are opened to send video upward i.e. for outgoing video streams
    2. 
+                    
  2. 10 RTP ports are opened to send audio upward i.e. for outgoing audio streams
    3. 
+                    
  3. 10 RTP ports are opened to receive video i.e. for incoming video streams
    4. 
+                    
  4. 10 RTP ports are opened to receive audio i.e. for incoming audio streams
    5. 
+                

+				

+					Maximum bandwidth used by each video RTP port (media-track) is about 1MB; which can be controlled using "b=AS" session description parameter values. In two-way video-only session; 2MB bandwidth is used by each peer; otherwise; a low-quality blurred video will be delivered.
+				

+				
+// removing existing bandwidth lines
+sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , '');
+
+// setting "outgoing" audio RTP port's bandwidth to "50kbit/s"
+sdp = sdp.replace( /a=mid:audio\r\n/g , 'a=mid:audio\r\nb=AS:50\r\n');
+
+// setting "outgoing" video RTP port's bandwidth to "256kbit/s"
+sdp = sdp.replace( /a=mid:video\r\n/g , 'a=mid:video\r\nb=AS:256\r\n');
+

+                

+                    Possible issues:

+                
    
+                    
  1. Blurry video experience
    2. 
+                    
  2. Unclear voice and audio lost
    3. 
+                    
  3. Bandwidth issues / slow streaming / CPU overwhelming
    4. 
+                

+                
Solution? Obviously a media server!

+            

+        

         
-		
-        
-        
+	
+        
+        
     
 
diff --git a/meeting/meeting.js b/meeting/meeting.js
index 8dd0769f..a80f2631 100644
--- a/meeting/meeting.js
+++ b/meeting/meeting.js
@@ -443,6 +443,7 @@
 
             peer.createOffer(function(sdp) {
                 peer.setLocalDescription(sdp);
+                if(isFirefox) config.onsdp(sdp, config.to);
             }, null, offerAnswerConstraints);
 
             function sdpCallback() {
diff --git a/meeting/simple.html b/meeting/simple.html
new file mode 100644
index 00000000..7bfd3149
--- /dev/null
+++ b/meeting/simple.html
@@ -0,0 +1,53 @@
+
+
+
+        
+
+    
+        
+    
+    
+        
+    
+

+            

+                Local Media Stream

+            

+        

+            

+                Remote Media Streams

+            

+        

+
+
diff --git a/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/index.html b/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/index.html
index 41ece8fc..32f0a834 100644
--- a/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/index.html
+++ b/part-of-screen-sharing/webrtc-and-part-of-screen-sharing/index.html
@@ -205,7 +205,6 @@ 

 
         

             Copyright © 2013 Muaz Khan<@muazkh>.
-                Firefox Nightly is recommended for this experiment.
         

         

             

@@ -364,10 +363,6 @@ 
Shared DIVs will be visible here ↓

 
             window.userID = Math.round(Math.random() * 60535) + 500000;
         
-        

-        

-
-        This experiment will fail on chrome out of their non-reliable channels implementation.
 
         

         

diff --git a/socket.io/README.md b/socket.io/README.md
index edcf933c..d478fc15 100644
--- a/socket.io/README.md
+++ b/socket.io/README.md
@@ -11,7 +11,7 @@ There is a built-in [Socket.io over Node.js](https://github.com/muaz-khan/WebRTC
 ```javascript
 // openSignalingChannel or openSocket!
 openSignalingChannel: function(config) {
-   var SIGNALING_SERVER = 'http://domain.com:8888/';
+   var SIGNALING_SERVER = 'http://webrtc-signaling.jit.su:80/';
    var channel = config.channel || this.channel || 'default-channel';
    var sender = Math.round(Math.random() * 60535) + 5000;
    
diff --git a/video-conferencing/README.md b/video-conferencing/README.md
index b8f24b94..1b33b193 100644
--- a/video-conferencing/README.md
+++ b/video-conferencing/README.md
@@ -1,63 +1,160 @@
 #### WebRTC Group video sharing / [Demo](https://www.webrtc-experiment.com/video-conferencing/)
 
+1. Mesh networking model is implemented to open multiple interconnected peer connections
+2. Maximum peer connections limit is 256 (on chrome)
+
 =
 
 #### How `video conferencing` Works?
 
-In simple words, `multi-peers` and `sockets` are opened to make it work!
-
-For 10 people sharing videos in a group:
+Huge bandwidth and CPU-usage out of multiple peers interconnection:
 
-1. Creating 10 `unique` peer connections
-2. Opening 10 unique `sockets` to exchange SDP/ICE
+To understand it better; assume that 10 users are sharing video in a group. 40 RTP-ports (i.e. streams) will be created for each user. All streams are expected to be flowing concurrently; which causes blur video experience and audio lose/noise (echo) issues
 
 =
 
-For your information; in One-to-One video session; 4 RTP streams/ports get open:
+#### For each user:
 
-1. One RTP port for **outgoing video**
-2. One RTP port for **outgoing audio**
-3. One RTP port for **incoming video**
-4. One RTP port for **incoming audio**
+1. 10 RTP ports are opened to send video upward i.e. for outgoing video streams
+2. 10 RTP ports are opened to send audio upward i.e. for outgoing audio streams
+3. 10 RTP ports are opened to receive video i.e. for incoming video streams
+4. 10 RTP ports are opened to receive audio i.e. for incoming audio streams
 
 =
 
-So, for 10 peers sharing video in a group; `40 RTP` ports get open. Which causes:
-
-1. Blurry video experience
-2. Unclear voice
-3. Bandwidth issues / slow streaming
-
-=
+Maximum bandwidth used by each video RTP port (media-track) is about 1MB; which can be controlled using "b=AS" session description parameter values. In two-way video-only session; 2MB bandwidth is used by each peer; otherwise; a low-quality blurred video will be delivered.
 
-The best solution is to use a **middle media server** like **asterisk** or **kamailio** to broadcast your camera stream. 
+```javascript
+// removing existing bandwidth lines
+sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , '');
 
-To overcome burden and to deliver HD stream over thousands of peers; we need a media server that should **broadcast** stream coming from room owner's side.
+// setting "outgoing" audio RTP port's bandwidth to "50kbit/s"
+sdp = sdp.replace( /a=mid:audio\r\n/g , 'a=mid:audio\r\nb=AS:50\r\n');
 
-Process should be like this:
+// setting "outgoing" video RTP port's bandwidth to "256kbit/s"
+sdp = sdp.replace( /a=mid:video\r\n/g , 'a=mid:video\r\nb=AS:256\r\n');
+```
 
-1. Conferencing **initiator** opens **peer-to-server** connection
-2. On successful handshake; media server starts broadcasting **remote stream** over all other thousands of peers.
-
-It means that those peers are not connected directly with **room initiator**.
-
-But, this is **video broadcasting**; it is not **video conferencing**.
+=
 
-In video-conferencing, each peer connects directly with all others.
+#### Possible issues:
 
-To make a **media server** work with video-conferencing also to just open only-one peer connection for each user; I've following assumptions:
+1. Blurry video experience
+2. Unclear voice and audio lost
+3. Bandwidth issues / slow streaming / CPU overwhelming
 
-1. Room initiator opens **peer-to-server** connection
-2. Server gets **remote stream** from room initiator
-3. A participant opens **peer-to-server** connect
-4. Server gets **remote stream** from that participant
-5. Server sends **remote stream** coming from participant toward **room iniator**
-6. Server also sends **remote stream** coming from room-initiator toward **participant**
-7. Another participant opens **peer-to-server** connection...and process continues.
+Solution? Obviously a media server!
 
-Media server should mix all video streams; and stream it over single RTP port.
+=
 
-If 10 room participants are sending video streams; server should mix them to generate a single media stream; then send that stream over single **incoming** RTP port opened between server and **room initiator**.
+#### Want to use video-conferencing in your own webpage?
+
+```html
+
+
+
+
+
+

+

+        
+
+```
 
 =
 
diff --git a/video-conferencing/conference.js b/video-conferencing/conference.js
index 8b8d7d8a..7d530677 100644
--- a/video-conferencing/conference.js
+++ b/video-conferencing/conference.js
@@ -1,16 +1,15 @@
-// 2013, @muazkh » github.com/muaz-khan
-// MIT License » https://webrtc-experiment.appspot.com/licence/
-// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing
+// Muaz Khan     - https://github.com/muaz-khan
+// MIT License   - https://www.webrtc-experiment.com/licence/
+// Documentation - https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing
 
 var conference = function(config) {
     var self = {
         userToken: uniqueToken()
     };
-    var channels = '--',
-        isbroadcaster,
-        isGetNewRoom = true,
-        sockets = [],
-        defaultSocket = { };
+    var channels = '--', isbroadcaster;
+    var isGetNewRoom = true;
+    var sockets = [];
+    var defaultSocket = { };
 
     function openDefaultSocket() {
         defaultSocket = config.openSocket({
@@ -229,9 +228,9 @@ var conference = function(config) {
         joinRoom: function(_config) {
             self.roomToken = _config.roomToken;
             isGetNewRoom = false;
-			
-			self.joinedARoom = true;
-			self.broadcasterid = _config.joinUser;
+
+            self.joinedARoom = true;
+            self.broadcasterid = _config.joinUser;
 
             openSubSocket({
                 channel: self.userToken
diff --git a/video-conferencing/index.html b/video-conferencing/index.html
index 6e2c6e43..b0911bdd 100644
--- a/video-conferencing/index.html
+++ b/video-conferencing/index.html
@@ -1,256 +1,368 @@
-
-
-
-
-    WebRTC video-conferencing experiment ® Muaz Khan
-    
-    
-    
-    
-    
-    
-    
+        
+        
+        
+        
+        
+        
+    
+
+    
+        

+            

+                

+                    WebRTC 
+                    » 
+                    video-conferencing
+                    ® 
+                    Muaz Khan
+                
            
+                

+                    Copyright © 2013
+                    Muaz Khan<@muazkh>
+                    »
+                    @WebRTC Experiments
+                    »
+                    Google+
+                    »
+                    What's New?
+                

+            

+
+            

+
+            

+                

+            

+        
+            
+            
                
+                

+                    
+                        Private ?? #123456789
+                    
+                    
+                    
+                    
+                

+                
+                
+                

+                
+                
+                

+            

+        
+            
-
-
-
-    
-	
-    

-        
Features

-
-        
    
-            
  1. Multi-user video sharing (many-to-many)
    2. 
-            
  2. Cross-browser interoperability
    3. 
-        

-
-	    

-        

-
-        You can use RTCMultiConnection.js for audio/video/screen sharing (i.e. conferencing/broadcasting); it supports multi-sessions establishment on the same page, group text chat and file sharing, auto user presence detection and many other features.
-			
-		

-        

-        

-                
Feedback

+                /* UI specific */
+                var videosContainer = document.getElementById('videos-container') || document.body;
+                var btnSetupNewRoom = document.getElementById('setup-new-room');
+                var roomsList = document.getElementById('rooms-list');
+
+                if (btnSetupNewRoom) btnSetupNewRoom.onclick = setupNewRoomButtonClickHandler;
+
+                function hideUnnecessaryStuff() {
+                    var visibleElements = document.getElementsByClassName('visible'),
+                        length = visibleElements.length;
+                    for (var i = 0; i < length; i++) {
+                        visibleElements[i].style.display = 'none';
+                    }
+                }
+
+                function rotateVideo(video) {
+                    video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(0deg)';
+                    setTimeout(function() {
+                        video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(360deg)';
+                    }, 1000);
+                }
+
+                (function() {
+                    var uniqueToken = document.getElementById('unique-token');
+                    if (uniqueToken)
+                        if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '
Share this link
';
+                        else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace( /\./g , '-');
+                })();
+                
+                function scaleVideos() {
+                    var videos = document.querySelectorAll('video'),
+                        length = videos.length, video;
+
+                    var minus = 130;
+                    var windowHeight = 700;
+                    var windowWidth = 600;
+                    var windowAspectRatio = windowWidth / windowHeight;
+                    var videoAspectRatio = 4 / 3;
+                    var blockAspectRatio;
+                    var tempVideoWidth = 0;
+                    var maxVideoWidth = 0;
+
+                    for (var i = length; i > 0; i--) {
+                        blockAspectRatio = i * videoAspectRatio / Math.ceil(length / i);
+                        if (blockAspectRatio <= windowAspectRatio) {
+                            tempVideoWidth = videoAspectRatio * windowHeight / Math.ceil(length / i);
+                        }
+                        else {
+                            tempVideoWidth = windowWidth / i;
+                        }
+                        if (tempVideoWidth > maxVideoWidth)
+                            maxVideoWidth = tempVideoWidth;
+                    }
+                    for (var i = 0; i < length; i++) {
+                        video = videos[i];
+                        if (video)
+                            video.width = maxVideoWidth - minus;
+                    }
+                }
+
+                window.onresize = scaleVideos;
+
+            
+            
+            

+                
    
+                    
  1. Mesh networking model is implemented to open multiple interconnected peer connections
    2. 
+                    
  2. Maximum peer connections limit is 256 (on chrome)
    3. 
+                

+            

+        
+            

+                
Latest Updates

+                

+            

+        
+            

+                
Feedback

                 

-                    
+                    
                 

-                
+                Enter your email too; if you want "direct" reply!
+            

+            
+            

+                
How it works?

+                

+                       Huge bandwidth and CPU-usage out of multiple peers interconnection:
+                   
+                

+                

+                    To understand it better; assume that 10 users are sharing video in a group.
+                    40 RTP-ports (i.e. streams) will be created for each user. All streams are expected
+                    to be flowing concurrently; which causes blur video experience and audio lose/noise (echo)
+                    issues.

+                

+                    For each user:

+                
    
+                    
  1. 10 RTP ports are opened to send video upward i.e. for outgoing video streams
    2. 
+                    
  2. 10 RTP ports are opened to send audio upward i.e. for outgoing audio streams
    3. 
+                    
  3. 10 RTP ports are opened to receive video i.e. for incoming video streams
    4. 
+                    
  4. 10 RTP ports are opened to receive audio i.e. for incoming audio streams
    5. 
+                

+                

+                    Maximum bandwidth used by each video RTP port (media-track) is about 1MB; which can be controlled using "b=AS" session description parameter values. In two-way video-only session; 2MB bandwidth is used by each peer; otherwise; a low-quality blurred video will be delivered.
+                

+                
+// removing existing bandwidth lines
+sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , '');
+
+// setting "outgoing" audio RTP port's bandwidth to "50kbit/s"
+sdp = sdp.replace( /a=mid:audio\r\n/g , 'a=mid:audio\r\nb=AS:50\r\n');
+
+// setting "outgoing" video RTP port's bandwidth to "256kbit/s"
+sdp = sdp.replace( /a=mid:video\r\n/g , 'a=mid:video\r\nb=AS:256\r\n');
+

+                

+                    Possible issues:

+                
    
+                    
  1. Blurry video experience
    2. 
+                    
  2. Unclear voice and audio lost
    3. 
+                    
  3. Bandwidth issues / slow streaming / CPU overwhelming
    4. 
+                

+                
Solution? Obviously a media server!

             

-    

-    

+        
-    
-    
-
-
+            

+        
+    
+        
+        
+    
+
\ No newline at end of file
diff --git a/webrtc-broadcasting/README.md b/webrtc-broadcasting/README.md
index 6c82caaa..cdcf8fdd 100644
--- a/webrtc-broadcasting/README.md
+++ b/webrtc-broadcasting/README.md
@@ -4,6 +4,8 @@ Participants can view your broadcasted video **anonymously**. They can also list
 
 This experiment is actually a **one-way** audio/video/screen streaming.
 
+=
+
 You can:
 
 1. Share your screen in one-way over many peers
@@ -14,30 +16,199 @@ You can:
 
 #### How WebRTC One-Way Broadcasting Works?
 
-It is a **one-to-many** audio/video/screen sharing experiment. However, only room initiator will be asked to allow access to camera/microphone because his media stream will be shared in one-way over all connected peers.
+1. Mesh networking model is implemented to open multiple interconnected peer connections
+2. Maximum peer connections limit is 256 (on chrome)
+
+=
+
+It is one-way broadcasting; media stream is attached only by the broadcaster.
+
+It means that, if 10 people are watching your one-way broadcasted audio/video stream; on your system:
 
-It means that, if 10 people are watching your one-way broadcasted video stream; on your system:
+1. 10 RTP ports are opened to send video upward i.e. outgoing video
+2. 10 RTP ports are opened to send audio upward i.e. outgoing audio
 
-1. 10 unique peer connections are opened
-2. Same **LocalMediaStream** is attached over all those **10 peers**
+And on participants system:
 
-Behind the scene:
+1. 10 RTP ports are opened to receive video i.e. incoming video
+2. 10 RTP ports are opened to receive audio i.e. incoming audio
 
-1. 10 unique RTP ports are opened for **outgoing local audio stream**
-2. 10 unique RTP ports are opened for **outgoing local video stream**
+Maximum bandwidth used by each video RTP port (media-track) is about 1MB. You're streaming audio and video tracks. You must be careful when streaming video over more than one peers. If you're broadcasting audio/video over 10 peers; it means that 20MB bandwidth is required on your system to stream-up (broadcast/transmit) your video. Otherwise; you'll face connection lost; CPU usage issues; and obviously audio-lost/noise/echo issues.
 
-So, total **20 RTP ports** are opened on your system to make it work!
+You can handle such things using "b=AS" (application specific bandwidth) session description parameter values to deliver a little bit low quality video.
 
-Also, **10 unique sockets** are opened to exchange SDP/ICE!
+```javascript
+// removing existing bandwidth lines
+sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , '');
 
-Remember, there is **no incoming RTP port** is opened on your system! **Because it is one-way streaming**!
+// setting "outgoing" audio RTP port's bandwidth to "50kbit/s"
+sdp = sdp.replace( /a=mid:audio\r\n/g , 'a=mid:audio\r\nb=AS:50\r\n');
 
-For users who are watching your video stream anonymously; **2 incoming RTP** ports are opened on each user's side:
+// setting "outgoing" video RTP port's bandwidth to "256kbit/s"
+sdp = sdp.replace( /a=mid:video\r\n/g , 'a=mid:video\r\nb=AS:256\r\n');
+```
 
-1. One RTP port for **incoming remote audio stream**
-2. One RTP port for **incoming remote video stream**
+=
+
+Possible issues
+
+1. Blurry video experience
+2. Unclear voice and audio lost
+3. Bandwidth issues / slow streaming / CPU overwhelming
+
+Solution? Obviously a media server!
+
+=
 
-Again, because it is one-way streaming; **no outgoing RTP ports** will be opened on room participants' side.
+#### Want to use video-conferencing in your own webpage?
+
+```html
+
+
+
+
+
+
+
+        
+

+

+        
+
+```
 
 =
 
diff --git a/webrtc-broadcasting/broadcast-ui.js b/webrtc-broadcasting/broadcast-ui.js
deleted file mode 100644
index ea7d4508..00000000
--- a/webrtc-broadcasting/broadcast-ui.js
+++ /dev/null
@@ -1,176 +0,0 @@
-// 2013, @muazkh » github.com/muaz-khan
-// MIT License » https://webrtc-experiment.appspot.com/licence/
-// Documentation » https://github.com/muaz-khan/WebRTC-Experiment/tree/master/webrtc-broadcasting
-
-var config = {
-    openSocket: function (config) {
-        var SIGNALING_SERVER = 'https://www.webrtc-experiment.com:8553/',
-            defaultChannel = location.hash.substr(1) || 'video-oneway-broadcasting';
-
-        var channel = config.channel || defaultChannel;
-        var sender = Math.round(Math.random() * 999999999) + 999999999;
-
-        io.connect(SIGNALING_SERVER).emit('new-channel', {
-            channel: channel,
-            sender: sender
-        });
-
-        var socket = io.connect(SIGNALING_SERVER + channel);
-        socket.channel = channel;
-        socket.on('connect', function() {
-            if (config.callback) config.callback(socket);
-        });
-
-        socket.send = function(message) {
-            socket.emit('message', {
-                sender: sender,
-                data: message
-            });
-        };
-
-        socket.on('message', config.onmessage);
-    },
-    onRemoteStream: function (htmlElement) {
-        htmlElement.setAttribute('controls', true);
-        participants.insertBefore(htmlElement, participants.firstChild);
-        htmlElement.play();
-        rotateInCircle(htmlElement);
-    },
-    onRoomFound: function (room) {
-        var hash = location.hash.replace('#', '').length;
-        if (!hash) {
-            var alreadyExist = document.getElementById(room.broadcaster);
-            if (alreadyExist) return;
-
-            if (typeof roomsList === 'undefined') roomsList = document.body;
-
-            var tr = document.createElement('tr');
-            tr.setAttribute('id', room.broadcaster);
-
-            if (room.isAudio) tr.setAttribute('accesskey', room.isAudio);
-
-            tr.innerHTML = '' + room.roomName + '' +
-                '';
-            roomsList.insertBefore(tr, roomsList.firstChild);
-
-            tr.onclick = function () {
-                tr = this;
-                broadcastUI.joinRoom({
-                    roomToken: tr.querySelector('.join').id,
-                    joinUser: tr.id,
-                    isAudio: tr.getAttribute('accesskey')
-                });
-                hideUnnecessaryStuff();
-            };
-        } else {
-            /* auto join privately shared room */
-            config.attachStream = null;
-            broadcastUI.joinRoom({
-                roomToken: room.roomToken,
-                joinUser: room.broadcaster,
-                isAudio: room.isAudio
-            });
-            hideUnnecessaryStuff();
-        }
-    },
-    onNewParticipant: function (participants) {
-        var numberOfParticipants = document.getElementById('number-of-participants');
-        if (!numberOfParticipants) return;
-        numberOfParticipants.innerHTML = participants + ' room participants';
-    }
-};
-
-function createButtonClickHandler() {
-    captureUserMedia(function () {
-        var shared = 'video';
-        if (window.option == 'Only Audio') shared = 'audio';
-        if (window.option == 'Screen') shared = 'screen';
-        broadcastUI.createRoom({
-            roomName: (document.getElementById('conference-name') || {}).value || 'Anonymous',
-            isAudio: shared === 'audio'
-        });
-    });
-    hideUnnecessaryStuff();
-}
-
-function captureUserMedia(callback) {
-    var constraints = null;
-    window.option = broadcastingOption ? broadcastingOption.value : '';
-    if (option === 'Only Audio') {
-        constraints = {
-            audio: true,
-            video: false
-        };
-    }
-    if (option === 'Screen') {		
-        var video_constraints = {
-            mandatory: {
-                chromeMediaSource: 'screen'
-            },
-            optional: []
-        };
-        constraints = {
-            audio: false,
-            video: video_constraints
-        };
-    }
-
-    var htmlElement = document.createElement(option === 'Only Audio' ? 'audio' : 'video');
-    htmlElement.setAttribute('autoplay', true);
-    htmlElement.setAttribute('controls', true);
-    participants.insertBefore(htmlElement, participants.firstChild);
-
-    var mediaConfig = {
-        video: htmlElement,
-        onsuccess: function (stream) {
-            config.attachStream = stream;
-            callback && callback();
-
-            htmlElement.setAttribute('muted', true);
-            rotateInCircle(htmlElement);
-        },
-        onerror: function () {
-			if (option === 'Only Audio') alert('unable to get access to your microphone');
-			else if(option === 'Screen') {
-				if(location.protocol === 'http:') alert('Please test this WebRTC experiment on HTTPS.');
-				else alert('Screen capturing is either denied or not supported. Are you enabled flag: "Enable screen capture support in getUserMedia"?');
-			}
-			else alert('unable to get access to your webcam');
-        }
-    };
-    if (constraints) mediaConfig.constraints = constraints;
-    getUserMedia(mediaConfig);
-}
-
-/* on page load: get public rooms */
-var broadcastUI = broadcast(config);
-
-/* UI specific */
-var participants = document.getElementById("participants") || document.body;
-var startConferencing = document.getElementById('start-conferencing');
-var roomsList = document.getElementById('rooms-list');
-
-var broadcastingOption = document.getElementById('broadcasting-option');
-
-if (startConferencing) startConferencing.onclick = createButtonClickHandler;
-
-function hideUnnecessaryStuff() {
-    var visibleElements = document.getElementsByClassName('visible'),
-        length = visibleElements.length;
-    for (var i = 0; i < length; i++) {
-        visibleElements[i].style.display = 'none';
-    }
-}
-
-function rotateInCircle(video) {
-    video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(0deg)';
-    setTimeout(function () {
-        video.style[navigator.mozGetUserMedia ? 'transform' : '-webkit-transform'] = 'rotate(360deg)';
-    }, 1000);
-}
-
-(function () {
-    var uniqueToken = document.getElementById('unique-token');
-    if (uniqueToken) if (location.hash.length > 2) uniqueToken.parentNode.parentNode.parentNode.innerHTML = '
Share this link
';
-    else uniqueToken.innerHTML = uniqueToken.parentNode.parentNode.href = '#' + (Math.random() * new Date().getTime()).toString(36).toUpperCase().replace(/\./g, '-');
-})();
\ No newline at end of file
diff --git a/webrtc-broadcasting/index.html b/webrtc-broadcasting/index.html
index 091af9c3..49204c28 100644
--- a/webrtc-broadcasting/index.html
+++ b/webrtc-broadcasting/index.html
@@ -1,218 +1,361 @@
-
-
-    
+
+
+
     
-        WebRTC One-Way Audio, Video or Screen Sharing/Broadcasting ® Muaz Khan
+        WebRTC Broadcasting ® Muaz Khan
         
-        
+        
         
         
         
         
+        
+        
+        
         
-        
         
+        
+        
+        
+        
+        
     
-    
+
     
         

-            ↑ WEBRTC EXPERIMENTS
+            

+                

+                    WebRTC 
+                    Broadcasting
+                    ® 
+                    Muaz Khan
+                
            
+                

+                    Copyright © 2013
+                    Muaz Khan<@muazkh>
+                    »
+                    @WebRTC Experiments
+                    »
+                    Google+
+                    »
+                    What's New?
+                

+            

 
-            

-				WebRTC one-way audio, video and screen sharing
-			

+            

 
-            

-				Copyright © 2013 Muaz Khan<@muazkh>.
-			

-			
-			

+            

                 

             

-			
-            
-                
-                    
-                    
-                    
-                
-            

-                        
-                    
-                        
                             
                             
                             
-                        
-                    
-                        
-                    

-            

-            
-                
-                    
-                
-            
 Private broadcast ??
-                        

-                            /webrtc-broadcasting/#123456789
-                        

-                    

-            

-            
-			
-            
-            
-            
-            
    
-                
  1. Share your screen in one-way over many peers
    2. 
-                
  2. Share you camera in one-way over many peers
    3. 
-                
  3. Share/transmit your voice in one-way over many peers
    4. 
-            

+                    
+                    
+                    
+                

+                
+                
+                

+                
+                
+                

+            

+        
+            
+            
+            

+                
    
+                    
  1. Mesh networking model is implemented to open multiple interconnected peer connections
    2. 
+                    
  2. Maximum peer connections limit is 256 (on chrome)
    3. 
+                

+            

+        
+            

+                
Latest Updates

+                

+            

+        
+            

+                
Feedback

                 

-                    
+                    
                 

-                
+                Enter your email too; if you want "direct" reply!
+            

+            
+            

+                
How it works?

+                

+                    It is one-way broadcasting; media stream is attached only by the broadcaster.
+                

+                

+                    It means that, if 10 people are watching your one-way broadcasted audio/video stream; on your system:
+                

+                
    
+                    
  1. 10 RTP ports are opened to send video upward i.e. outgoing video
    2. 
+                    
  2. 10 RTP ports are opened to send audio upward i.e. outgoing audio
    3. 
+                

+                

+                    And on participants system:
+                

+                
    
+                    
  1. 10 RTP ports are opened to receive video i.e. incoming video
    2. 
+                    
  2. 10 RTP ports are opened to receive audio i.e. incoming audio
    3. 
+                

+                

+                    Maximum bandwidth used by each video RTP port (media-track) is about 1MB. You're streaming audio and video tracks. You must be careful when streaming video over more than one peers. If you're broadcasting audio/video over 10 peers; it means that 20MB bandwidth is required on your system to stream-up (broadcast/transmit) your video. Otherwise; you'll face connection lost; CPU usage issues; and obviously audio-lost/noise/echo issues.
+                

+                

+                    You can handle such things using "b=AS" (application specific bandwidth) session description parameter values to deliver a little bit low quality video.
+                

+                
+// removing existing bandwidth lines
+sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , '');
+
+// setting "outgoing" audio RTP port's bandwidth to "50kbit/s"
+sdp = sdp.replace( /a=mid:audio\r\n/g , 'a=mid:audio\r\nb=AS:50\r\n');
+
+// setting "outgoing" video RTP port's bandwidth to "256kbit/s"
+sdp = sdp.replace( /a=mid:video\r\n/g , 'a=mid:video\r\nb=AS:256\r\n');
+

+                

+                    Possible issues:

+                
    
+                    
  1. Blurry video experience
    2. 
+                    
  2. Unclear voice and audio lost
    3. 
+                    
  3. Bandwidth issues / slow streaming / CPU overwhelming
    4. 
+                

+                
Solution? Obviously a media server!

             

         

         
-        
+    
+        
+        
     
-
 
\ No newline at end of file