Adds callstatsname to the permanent analytics properties.

ref(JitsiLocalTrack): get rid of 'setMutedInProgress'

The 'setMutedInProgress' flag is unreliable and it can lead to weird
situations when mute operation is in progress, but the removal from
JingleSessionPC is waiting on the queue. In such case LocalSdpMunger
will fake SDP even if the track is in the PeerConnection which may
confuse some parts of the SDP transformation chain.

For example RtxModifier will not attempt to inject Rtx group description
if it sees that the SSRC are already there. But it can happen that even
though the SSRCs are there they do not contain valid track ID, because
the track has changed.

The LocalSdpMunger is supposed to fake the SDP only when the video
MediaStream is not in the PeerConnection. So this commit implements just
that: a way to tell if JitsiLocalTrack's stream is currently in
the PeerConnection (added through TraceablePeerConnection).

[RN] Fixes problem with CallStats on mobile

There are whole objects logged and the react-native app would crash when
logging too much stuff.

As opoosed to using window.location, which is not available on all platforms,
notably on mobile.

Reduce the source code by replacing this.options.config with config.

It's no longer needed, and it hasn't been for a while.

Fire status message update when there is no status message.

In case there was a status message and later that message is removed from the presence, fire an event so the UI can also clear it.

via JSON messages received via the Chatroom

Random coding style-related issues that bothered me while reading the
source code touched by "Fix local video disappearing from local
thumbnail" (https://github.com/jitsi/lib-jitsi-meet/pull/571). These
include naming, formatting, comments, typos, duplication.

Commit "fix(JitsiLocalTrack): restore 'inMuteOrUnmuteProgress'" removed
all read occurrences of dontFireRemoveEvent so the one and only write
left is now obsolete.

Flag 'dontFireRemoveEvent' was unreliable, because if
the JitsiLocalTrack is muted/unmuted rapidly it's state can be false
at the time when ended event is triggered. The event handlers must be
unregistered to handle this gracefully.

Restores 'inMuteOrUnmuteProgress' flag used by the LocalSdpMunger.

(cherry picked from commit 2e6c4dd)

fix(JitsiConference): reduce verbosity level

There are scenarios when it's ok to call setP2PStatus again with
the same value. For example when P2P is stopped, before it starts with
the A/B testing mode enabled. It's only important to know that it
happened, but it's not an error, because the code will not execute
and return immediately.

* feat(quality-slider): expose video constraint method and data channel opening

* squash: rename to data channel opened

'echoCancellation' should be disabled by 'disableAEC'
'disableAP' should disable all options

Instead of failing with an error if another operation is in progress, which is
impossible to know as a user, chain them using the primises API.

fix(JSPC): local tracks not advertised

fix: never miss 'ondatachannel' event

Moves the call to 'setLocalDescription' from 'sendSessionInitiate' to
'invite' as it was easy to overlook.

In case when the answer is being set for the first time, full sRD/sLD
cycle is required to have the local description updated and SSRCs
synchronized correctly. Otherwise SSRCs for streams added after invite,
but before the answer was accepted will not be detected. The reason for
that is that renegotiate can not be called when adding tracks and they
will not be reflected in the local SDP.

The 'ondatachannel' even listener must be set as soon as
the TraceablePeerConnection is created and before the offer is accepted.
We've observed situations where the ondatachannel event was missed,
because the listener is bound too late from the JingleSessionPC
'acceptOffer' success callback.

Send resolution (in pixels) to the analytics backend.

As new use case for joining a conference without calling GUM has been
added to jitsi-meet enumerate devices should not be delayed. In fact not
only it was delayed, but never called in such case
(getUserMediaStatus.initialized stays false forever).

Fixes crash on undefined var in case get user media was not called,
before the conference started.

* make audio processing constraints configurable

AP: echoCancellation is a "master switch" for disabling all audio processing.  set to disableAP=true and stereo=true allows fullband, stereo sound.
AGC: misbehaves with certain 'pre-gained' mics.
HPF: can't hear any effect but it's the last constraint to make configurable.

* Update RTCUtils.js