* Updates recording dialog.

* Update config.js doc.

* Adds comment and make a check more intuitive.

* Changes of using enum for recording types.

The android.telecom.CallAudioState class was only added in API level 23 (Android
M), so make sure we don't import it in lower versioned devices.

* feat(screenshare): emit source type when starting screenshare

* squash: update doc

Bring over two fixes for spot. One is for
identifying the screenshare type when using
a camera for screenshare or when using a proxy
stream. Also bring in a fix to avoid a js error
in chrome ios.

* Adds integrations doc.

Google, Microsoft and Dropbox for now.

* Updates doc.

Seems like windows does not have emojis for flags.

* Adds new format of phoneList service and re-design dial in numbers page.

Adds flags and country names (with translations) for the numbers if using the new format.

* Fixes tests and fixes get default number.

* Updates swagger with new format.

* Moves html back yo table.

Fixes displaying on mobile and also the tel: URI generation. The tel: URI is tested on Android and iOS and seems to work (Android was not interpreting 'p', but both seems to like ',').

* Fixes a wrong return statement.

* Small fixes.

Update api.md

When a conference is started, the currently used
camera device id is saved. I believe this is happening
because lib-jitsi-meet does not use exact device id
mathcing when calling getUserMedia, so it's possibl
to request camera A but get camera B back because
camera A is not available. When config.startScreenSharing
is true, the syncing occurs and saves the desktop
source id. So when screensharing is stopped, jitsi-meet
requests that desktop source id instead of the preferred
camera.

Adds a new event fired when Jitsi Meet shows the feedback prompt.

The conference disconnection process is asynchronous which means there's
no guarantee that there will be CONFERENCE_LEFT event for the old
conference, before the next conference is joined. Because of that we can
end up with two simultaneous calls on the native side which is not
always supported. End the call on CONFERENCE_WILL_LEAVE to fix this
corner case.

Using an array of styles in Firefox causes an error
that triggers jitsi-meet to redirect to a static page.

They generate a bunch of harmless yet confusing error messages and they are not
really useful.

RTCAudioSession is a thin wrapper around AVAudioSession provided by the WebRTC
framework. It makes some use-cases easier, and leads us closer to manual audio
unit management, which we will likely need in the near future.

If a new conference is joined and as a result the current one is terminated,
don't ever attempt to set the audio mode.