Pull initial offer/answer tasks out of the modification queue and execute them right away. Only track and codec related operations that necessitate a renegotiation cycle need to be pushed to the modification queue. The queue execution is paused until the session is established. This avoids track operations being executed before the session is established. This fixes an issue seen in p2p connections where sources of the initiator are not signaled to the remote since the tracks are added while the initiator is waiting for a session-accept from the peer.

s/updateCurrentKey/updateCurrentMediaKey/g

* fix(stats) Obtain resolution/fps from 'outbound-rtp' stats.
'Track' based stats were dropped in Chrome 112, therefore send resolution and fps for the simulcast case needs to be calculated based on the 'outbound-rtp' streams that are currently active.

* squash: remove an unwanted log

* Squash: Address review comments.
Use outbound-rtp stats for both Firefox and Safari.
Firefox - Ignore active field if not present in the stats and calc fps using 'framesSent'.

With visitors we rarely see not able to send messages or presence because not connected error.
Seems the status of the connection is wrong when quickly disconnecting and connecting again.

Fixes an issue on mobile where the videos of users joining after the mobile endpoint are not being rendered.

Track based stats have been removed recently and RTCRtpReceiver:getSynchronizationSources has been available in the browsers for a while now.

Also add all the orphan jvb tracks to the conference when the call switches over from p2p to jvb.

Any jibri recording. streaming or sip jibri will join the main room.

When receiving error like max occupants reached we have not joined the room, and there is no point of sending presence unavailable and check and clean after 5 secs.

When configureSenderVideoEncodings task is pushed to the modification queue, there might not be a setParameters call in progress and this new one will not get chanined to any promise. However, when the task finally is executed, there is a chance for another setParameters method execution to be in progress and can result in setParameters call failing. Also, these extra steps to ensure that the sender encodings are correctly encoded are no longer needed since with multi-stream mode, we do not change the source video type and the previous encodings config will still be valid.

* Add a feature flag and xmpp-caps entry for joining as a visitor.

* Default to enabled, use for the confernece-request flag too.

* fix(TPC) Avoid applying constraints on desktop tracks if not needed.
Ignore sender constraints if the client is already sending video of the requested resolution. For desktop tracks, max resolution will be the height of the window being captured irrespective of the height being requested by the peer. Therfore, check if the configured resolution is equal to the track height for all requested heights > 0.
Fixes an issue where the track addition fails because of setParameters failing on the video track. This is only seen the torture tests because this is very timing specific.

* squash: check correct scaleResolutionDownBy value for p2p conn.

* feat(visitors): Visitors chat.

For messages coming from visitors, which are not in the current room we need to supply a display name.
Mark messages with custom display name as coming from guest.

* feat: Adds a visitors version to conferenceIQ.

Request higher capture resolutions when desktopSharingFrameRate is not defined in config.js.

Update notes:

- The base64-sha1 code got stripped out, so we bundle it now
- There are 2 optional dependencies which we want to ignore to avoid
  extra bundle size: xmldom and ws (the former is used by RN but we
  provide it in the app and the latter is only used by Node
  environments)

* feat: Add initial-last-n to jingle if config.channelLastN is set.

* Also add initial-last-n to transport-accept.

This fixes an issue where p2p clients (with different codec preferences) fail to decode video because the negotiated codecs are removed from the supported codecs list after the media session is established. The codec preferences will be applied when the first offer/answer is created.

This should avoid unnecessary renegotiations if disabledCodec setting is present in config.js and should improve time to receive media when joining big calls.

* fix(TPC) Disable media instead of changing dir for p2p->jvb switch.
Resume or suspend the media on the jvb peerconnection by changing the RTCRtpEncodingParamters.active state instead of changing the direction on the transceiver. This avoids the needs to start a O/A renegotiation cycle for these operations. The media direction will be changed only for p2p lastn=0 case since video needs to be disabled on both the sender and the peer for p2p lastn=0 case.

* Address review comments

* Disable media after adding source while media is suspended on the jvb connection. Default 'active' state for stream encodings after the source is added is 'true'.

* Wait for all the promises to be settled before returning

Currently, the client doesn't attempt to re-establish bridge WS when the connection is closed by the remote end with a code 1001. We have noticed that Cloudflare terminates the WS with the same error code when it recycles its proxies. When that happens, client ends up not having a bridge channel which can result quality issues.
Client now tries to re-establish connection when there are remote endpoints in the call.

* fix(TPC): make screen share bitrate configurable

* fix(TPC): make local stream encodings configurable

* fix(TPC): fix lint error

* fix(TPC): alpha sort methods and update check for VP8

* feat(ssrc-rewriting) Advertise support to Jicofo.

* Fix unit test

* fix(SignalingLayer) Update SSRC owners on leave.
Update the SSRC owners in the following cases:
1. When a remote endpoint leaves the call.
2. When a source-remove is received.
3. When a source is remapped (with ssrc-rewriting enabled).
Create the remote track even if presence is not yet received. The ssrc owner check prevents the client from creating a dummy track when the call switches over from p2p to jvb when the last remote endpoint leaves the call.

* ref(SignalingLayer) alpha sort methods.
Clean up unused methods, _findEndpointSourceInfoForMediaType is not used anymore.

* squash: Address review comments.

When the application doesn't set receiver constraints, generate default constraints in the source-name mode.
Cleanup code related to the endpoint based video constraint handling for p2p connection.

Endpoint based receiver constraints and other endpoint based bridge signaling messages are no longer supported by latest JVB after the switch to source-name signaling.
Rename method names 'sendNewReceiverVideoConstraintsMessage'->'sendReceiverVideoConstraintsMessage', 'setNewReceiverVideoConstraints'->'setReceiverVideoConstraints'

Simulcast for SS is always on in unified plan mode, the lower spatial resolution streams are turned off for low fps sharing.

It was behind a testing flag setScreenSharingResolutionConstraints.