RTCMultiConnection-v1.4 released for early tests

It is experimental releases. Following features are excluded from this
release:

1. Streams Renegotiation
2. Auto Close Entire Session
3. Extra Data

All missing features will be added in next commits.

In this release:

1. You can us any signaling gateway for signaling; no dynamic channels
or namespaces are required. XHR POST/GET based signaling also worsk.
SIP-over-WebSockets too.

https://github.com/muaz-khan/WebRTC-Experiment/tree/master/video-conferencing#custom-signaling-channel

2. You can set "frame rate" for audio streams like this:

connection.framerate = {
--- minptime: 10,
---  maxptime: 60
};

3. You can get alerted if a stream stopped:

connection.onstreamended = function(e) {
--- document.body.removeChild(e.mediaElement);
};

4. You can stop any stream:

connection.streams['stream-id'].stop();

https://webrtc-experiment.appspot.com/RTCMultiConnection-v1.4.js