'onmute'/'onunmute' event of MediaStreamTrack can be used to detect that
the remote user is having connectivity issues, because no video data is
received. We check if those are in sync with the signalling and if not
trigger connection interrupted updates with small delay.

Add "onmute" and "onmute" handlers to the WebRTC MediaStream and trigger
corresponding RTCEvents.

The event is triggered when the notification about the user's connection
status change arrives from the JVB.

Fixes resolving this when passing function after enabling es6.

This currently fixes problem for participants joining in a room with other participants and trying to send initial value of selected local participant will throw an error. And fixes a potential problem if the UI can select participants before their video is available, before session initiated, can make local state with jvb inconsistent. Also if we want to have multiple selected endpoints or pinned participants and this is done before session-initiate caching should be done before creating data channel (currently selecting multiple is not possible).

Finds correct error when getUserMedia succeed with only one of the audio or video, and the other fails.

Fixes the availableDevices, to reflect the devices for which we succeeded or failed to obtain stream.

Stores render event in connection times to use it for tests.

Moves render time calculations in the library.

On Safari/IE we have to wait until the WebRTC stack is
initialized in case the plugin is being installed for
the first time.