Handle the promise returned by RTCRtpSender.setParameters.

(cherry picked from commit a14b3c80ae)

* ref: Moves xmpp logs to be accessed from connection.

In cases where there is no room like pre-join and lobby screen we still want to be able to debug xmpp messages.

* squash: Remove getLogs from Conference.

* squash: Fix catch.

* squash: Print the error.

When the client starts video muted and the track is later added, make sure we construct the simulcasr stream constraints then.

When the video sender constraint changes in the p2p case when the remote user switches between stage/tile view,
get a downsampled video of the original 720p stream instead of requesting the camera for a new stream with the lower resolution.
Fixes https://github.com/jitsi/jitsi-meet/issues/7267

* feat: Apply max bitrates on video sender
Add the ability to control the max bitrates on the video sender through a config.js videoQuality setting.

* ref: Use '_' prefix only for internal methods

Add a performance stat around long tasks. Chrome supports PerformanceObserver API that lets us
register for long tasks event. Any task that takes longer than 50ms is considered a long task.

Bumps [elliptic](https://github.com/indutny/elliptic) from 6.5.2 to 6.5.3.
- [Release notes](https://github.com/indutny/elliptic/releases)
- [Commits](https://github.com/indutny/elliptic/compare/v6.5.2...v6.5.3)

Signed-off-by: dependabot[bot] <support@github.com>

The audio levels reported on the audio receivers are lower when compared to the value reported by getStats.
Values reported by getStats on chrome do not follow the the spec and since we have combination of clients using both getStats and getSynchronizationSources,
lets stick to one scale to make them look uniform.
Also, the receivers seem to be reporting audio level for a little bit after the remote user has muted. Make sure the track is unmuted
before setting the audio level on the track.

An 'item-not-found' or other error on Websocket reconnect will refresh
the timestamp which will give misleading information about when
the last stanza was received.

Updates strophejs-plugin-stream-management
to 001cf02bef which fixes
few corner cases around XMPP WebSocket reconnect logic.

With the introduction of lobby the ChatRoom KICKED event is now also emitted for
ourselves (the kicker) so we want to avoid emitting an event where `undefined`
kicked someone.

Remove redundant _channelOpen flag from RTC and rely on the channel's isOpen() getter instead.
This fixes an error throw when the _channelOpen flag was not updated while the channel's been
closed by the bridge.

Delays e2e encryption initialization until any encryption key is set.
Recreates peerconnections in order to enable insertable streams
only if the e2ee is used. This is to avoid bug around insertable streams
which may cause audio issue when the main JavaScript thread is loaded:

https://bugs.chromium.org/p/chromium/issues/detail?id=1103280

Moves the logics which deal with JitsiConference events to a separate module.

JitsiConference should emit JitsiConference events instead of RTC events
when adding/removing tracks on jvb<->p2p switch. This behaviour was mixing RTC layer
with the conference one. When conference switches from jvb to p2p it wants to make
the application switch to use p2p JitsiTracks, but it doesn't mean that the JVB tracks
have been removed on the RTC/peer connection level.

* feat: use getSynchronizationSources on the receiver for remote audio levels
Use getSynchronizationSources if it is supported, fallback to using getStats otherwise.
* feat/ref: Use the local audio levels from LocalStatsCollector
When using getSynchronizationSources, use the audio levels from LocalStatsCollector for NoAudioSignalDetection.js
Remove obsolete code - TalkMutedDetection feature using audio levels is not used anymore

* fix: Takes disabled encodings into account when calculating local resolution.
* feat: Adds a new event that's triggered when the max enabled resolution changes.
* feat: Broadcasts the max enabled resolution value along with other stats.

It's not yet supported, so assume success.

* fix(safari): override min/max video constraints on safari
Override the constraints on Safari because of the following webkit bug.
https://bugs.webkit.org/show_bug.cgi?id=210932
Camera doesn't start on older macOS versions if min/max constraints are specified.

Used for extending owner privileges to other participants in the chat room.

The same logic should apply to all Chromium based browsers, not just Chrome.

This is useful for testing insertable streams related issues.

This was an experimental feature introduced for calculating the RTT to STUN servers, no longer needed.

Adds extra 15 seconds before emitting ICE failed event. If the OS is capable of
reporting internet offline status then will start counting when the internet
comes back online.

The reason for that is when ICE restarts are disabled a user will get the page reload
screen immediately, but there's a chance to recover as long as the channels have not
expired on the bridge.

Add API to pass internet online/offline status to LJM
which allows it to make more informed decisions about
reconnects.

* ref: Fix imports order.

* feat: Adds display name required connection event.

* feat: Set degradation preference on video senders
Use 'maintain-framerate' for camera tracks and 'maintain-resolution' for desktop tracks

* ref: apply cached video contraints and degradation preference on unmute
Since the sender encodings are available only after the renegotiation is complete (on chrome/safari), constraints/preferences have to be applied after renegotiation is complete.