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

* 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.

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.

That's a graceful close by the bridge.

Reorganized the logic a bit to make it more clear.

1. Checks peer's preferred codec in p2p case. Mobile and web have different preferred codecs.
2. Log an error message when the preferred codec is not offered by JVB.
3. Clean up code related to deprecated config.js settings 'preferH264' and 'disableH264'.
4. Refactor the codec selection logic so that correct codec is picked.

It now supports the sender encodings API.

* fix(BridgeChannel): prevent multiple websockets on retries

* fix(BridgeChannel): ignore different websocket close event

* fix(BridgeChannel): Keep onclose log

* ref: Add comment for retry timeout on connecting ws

* ref: Use logger debug instead info

(Rather than moving streams to inactive in SDP.)

Revert this change as this is causing a dummy track to be created when the media resumes on jvb connection after the p2p peer has left the call.

Currently, remote tracks are not created if presence for the endpoint is not received before source signaling. With ssrc-rewriting, the source information will be received on the bridge channel and presence on the prosody ws so there are chances that they can be out of sync. We do not want to skip remote track creation when that happens.

It's emitted when remotely closed / failed.

* Remove support for legacy endpoint based signaling and make source-name signaling as the only signaling mode. The legacy screensharing mode will no longer be supported. The client will still be able to process remote presence sent in the old format to support interop with very old mobile clients and jigasi that do not support source-name signaling.
* remove code related to presenter mode.

Allow users to seamlessly switch which tab they are sharing without having to stop the current share and select a new tab again. This is supported on Chrome 107 onwards.

Instead of Jicofo signaling all the remote sources available in the call, the bridge now signals only a limited set of SSRCs and then rewrites the SSRC on the outgoing media streams. The SSRC mapping is done based on the sources requested by the clients through the receiver constraints. This limits the number of m-lines in the remote/local SDPs on the client and therefore results in better performance in large calls.

* Handle source remapping messages from bridge
* Added track_owner_changed events
* don't process an invalid rtx ssrc.
* keep track of remote ssrcs, only renegotiate on new ones.
* Change source name on remote track on ssrc remapping.
* Don't remove tracks on member leave.
* Remove (orphaned) tracks on session terminated.
* Use serial number (per media type) to create msid attribute.
* Update videoType on remapping.

Co-authored-by: James A <jqdrqgnq@users.noreply.github.com>

* fix(codec-selection) Munge all related m-lines for multi-stream.
Munge all the video m-lines to apply the preferred codec settings in multi-stream mode.

* fix(TPC) Cache transceiver mids for all local video tracks.
There will be multiple video tracks in multi-stream, we need to cache the mids for all the the associated transceivers.

* fix(sender-constraints) Do not apply sender constraints if value hasn't changed for a given source.

* fix(TPC) Configure the direction correctly for all m-lines.
Ignore negotiationneeded events for p2p to avoid going into a renegotiation loop.

* squash: Address review comments.

Fixes https://github.com/jitsi/jitsi-meet/issues/12319.

Adds a helper function getSourceIndexFromSourceName to
have track index parsing in one place.

In _processLocalSSRCsMap there was an assumption that a track
index can be a single digit only.

Use "last index of" or search from the end of the collection to
make parsing a bit more universal (does not matter as much for
Jitsi endpoint IDs, but if fixes issues if someone has
a customization there).

Firefox lists stopped transceivers when getTranceivers is called, ignore these when picking a trancceiver for a screenshare track. Fixes an issue when starting screenshare fails if there are stopped tranceivers in the peerconnection (i.e., if some remote users with sources have left the call).

Fixes cases where SS fails in multi-stream mode with p2p->jvb->p2p switching.

In Unified-plan mode, we rely on the browser to fire 'negotiationneeded' event to start a O/A cycle. Also add muted tracks to PC. Fixes issues where the streams are not being signaled when call switches over from p2p to jvb in multi-stream mode.

Fixes cases where SS stream is not being sent when p2p->jvb switch happens.

> Possibly fixes a bug where SS appears black when the'unmute' and forwarded sources event for the track are received before a large-video update is scheduled.

This fixes an issue where the bitrates for screenshare were much higher than before for VP9 causing the JVB to suspend SS streams more often.

It does no harm on RN, since it doesn't detect it as a supported browser
and bails out.

It gets resolved during translation.

Fixes: https://github.com/jitsi/lib-jitsi-meet/issues/2055

Since Track effects and mute/unmute operations use the same flow, i.e., removing/adding the track from the RTCPeerConnection but not from TPC, make the names of the methods involved more generic.

Since the track is only temporarily removed and added back, it can be treated like mute & unmute operation. Fixes https://github.com/jitsi/lib-jitsi-meet/issues/2048. Remove a TODO thats no longer needed, track replace operation don't force renegotiations anymore.

* feat: Add videoType to sources signaled in jingle.