video conferencing auto-join option removed

https://webrtc-experiment.appspot.com/video-conferencing/
Pre-recorded media streaming

Streaming pre-recorded video (media file)

Currently, using Firebase for streaming chunks of data because
MediaSource APIs are only supported on chrome canary which has
unreliable RTP (RTCDataChannel) streams.

Streaming WebM files only (in the moment!)

WebM file's size must be less than 1000KB; otherwise it will fail. It is
a bug will be fixed soon.

Demo:
https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/Pre-recorded-Media-Streaming/

Source code:
https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Pre-recorded-Media-Streaming
New Part of Screen Sharing Demo added

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/part-of-screen-sharing/
issues in Part of screen sharing demo recovered

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/part-of-screen-sharing/
Small changes

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/part-of-screen-sharing/
Part of screen sharing using RTCDataChannel

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/part-of-screen-sharing/RTCDataChannel/
https://webrtc-experiment.appspot.com/licence/

MIT License (MIT) included:
https://webrtc-experiment.appspot.com/licence/

Now, MIT will be the default licence for all future WebRTC Experiments.
Upding RecordRTC: http://bit.ly/RecordRTC-Demo

http://bit.ly/RecordRTC-Demo

https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC
RecordRTC: http://bit.ly/RecordRTC-Demo
RecordRTC can save longer videos without crash

Demo: http://bit.ly/RecordRTC-Demo
Source: http://bit.ly/RecordRTC-Demo

To use your own socket.io implementation; use this code:

var config = {
openSocket: function (config) {
var socket = io.connect('http://your-site:8888');
socket.channel = config.channel || 'WebRTC-Experiment';
socket.on('message', config.onmessage);

socket.send = function (data) {
socket.emit('message', data);
};

if (config.onopen) setTimeout(config.onopen, 1);
return socket;
}
};
RecordRTC possibilities of failures added

When saving recorded audio/video stream to disk, RecordRTC will fail if:

1. You're using chrome incognito mode
2. RecordRTC created duplicate temporary file
3. the requesting scheme is none of the following: http, https, chrome,
extension's, or file (only works with --allow-file-access-from-files)
4. the browser cannot create/initialize the metadata database for the
API under the profile directory
A few experiments documented .. for "how it works"

Updated RecordRTC, Video Conferencing, Group File Sharing, P2P File
Sharing, One-to-One video chat using socket.io, One-to-one video chat
using WebSockets etc.

Answered questions like "How", "Why" and "What". Other experiments will
be documented soon.
WebRTC Part of Screen Sharing updated

Share part of screen using RTCDataChannel:

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/part-of-screen-sharing/RTCDataChannel/

Sharing part of screen using Firebase:

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/part-of-screen-sharing/
Missed hangout.js file in part of screen sharing

Otherwise, this experiment may fail in your site.
RTCDataConnection library added

https://bit.ly/RTCDataConnection

RTCDataConnection is a library sits top of RTCPeerConnection and
RTCDataChannel to bring easiest file and data sharing experience.
RTCDataConnection auto-handles multi-user data connectivity like file
broadcasting or group file sharing.

Demos using RTCDataConnection:

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/RTCDataConnection/
RTCMultiConnection: bit.ly/RTCMultiConnection

http://bit.ly/RTCMultiConnection

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/RTCMultiConnection/demos/

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/RTCMultiConnection/

RTCMultiConnection highly simplifies multi-user connectivity along with
multi-session establishment.

https://github.com/muaz-khan/WebRTC-Experiment/wiki/RTCMultiConnection
DataChannel library, RTCMultiConnection, and more

http://bit.ly/DataChannel-Demo
RTCMultiConnection-v1.1.js library added.

RTCMultiConnection's syntax is simple to use. It is useful in each and
every WebRTC application.

It can help you write file sharing applications along with screen
sharing and audio/video conferencing applications — in minutes.

1. You can open multi-sessions and multi-connections — in the same page
2. You can share files of any size — directly
3. You can share text messages of any length
4. You can share one-stream in many unique ways — in the same page

Demos:

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/RTCMultiConnection-v1.1/

Documentation:
http:/bit.ly/RTCMultiConnection-Documentation
Typo fixed.
RTCPeerConnection-v1.5.js, DataChannel.js, RTCMultiConnection.js, and more

RTCPeerConnection-v1.5.js, DataChannel.js, RTCMultiConnection.js, and
more

WebRTC DataChannel and Firefox:

https://github.com/muaz-khan/WebRTC-Experiment/wiki/WebRTC-DataChannel-and-Firefox#points

RTCPeerConnection-v1.5:
https://webrtc-experiment.appspot.com/video-conferencing/
https://webrtc-experiment.appspot.com/chat-hangout/
https://webrtc-experiment.appspot.com/file-hangout/

RTCMultiConnection:

https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/RTCMultiConnection-v1.1

DataChannel:
https://googledrive.com/host/0B6GWd_dUUTT8RzVSRVU2MlIxcm8/DataChannel

Actually, Firefox changed their old RTCDataChannel implementation.
http://mozilla.github.com/webrtc-landing/DataChannel_changes.html

DataChannel.js supports fallback to websockets/socket.io or your
preferred signaling method.