MultiRTC, DataChannel.js, and few other demos updated.

https://www.webrtc-experiment.com/RTCMultiConnection/MultiRTC/
RTCPeerConnection-v1.6 added. Chat-Hangout and File-Hangout updated.
A few documents updated.
DetectRTC.js added: A library to detect WebRTC features!

https://www.webrtc-experiment.com/DetectRTC/
MultiRTC demo is now using WebSockets-over-Nodejs

NPM package: https://www.npmjs.org/package/multirtc
Demo: https://www.webrtc-experiment.com/RTCMultiConnection/MultiRTC/
RTCMultiConnection-v1.7, RecordRTC-Developement, MultiRTC and a few other updates.

http://www.rtcmulticonnection.org/changes-log/#v1.7
html2canvas, recordrtc, rtcmulticonnection-v1.7 etc. updated.

RecordRTC solves audio-sync issues by firing "onAudioProcessStarted".
HTML2Canvas fixes the always scroll-up issue. via #204
Html2Canvas.js is updated to support embedded <video> snapshots.

https://github.com/muaz-khan/WebRTC-Experiment/tree/master/part-of-screen-sharing

RTCMultiConnection-v1.7 is also improved and updated.
RecordRTC.js and relevant demos updated.
RTCMultiConnection-v1.8 added. Pluginfree Screen Sharing experiment updated. tabCapture chrome extension added.
A few minor updates regarding screen capturing and RTCMultiConnection-v1.8

https://github.com/muaz-khan/WebRTC-Experiment/tree/master/Chrome-Extensions/desktopCapture
RecordRTC-to-Socketio demo fixed and ffmpeg-logs-improved. Fixed #212

https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RecordRTC/RecordRTC-over-Socketio
RTCMultiConnection-v1.8 has a few new "useful" features!

http://www.rtcmulticonnection.org/changes-log/#v1.8
navigator.customGetUserMediaBar.js added. RTCMultiConnection-v1.8 updated.

https://www.webrtc-experiment.com/navigator.customGetUserMediaBar/
http://www.rtcmulticonnection.org/changes-log/#v1.8
screen.js, RTCMultiConnection-v1.8, part-fo-screen demo, updated.
RTCMultiConnection-v1.8, hark, customGetUserMediaBar, & a few demos updated.
RecordRTC breaking updates: Videos tracks are auto-skipped for audio recordings.

You must always pass {type:video} for audio+video (i.e. webm) recording;
otherwise pass {type:audio} to record only audio in ogg format.
RecordRTC will skip video tracks for ogg recordings.

// stream.getVideoTracks().length == 1
// stream.getAudioTracks().length == 1

// recording only audio
var audioRecorder = RecordRTC(stream, {
---- type: 'audio'
});

// recording audio+video
var videoRecorder = RecordRTC(stream, {
---- type: 'video'
});