Fixes an issue where media gets sent on jvb connection even when the connection is suspended.

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

When the call switches over to JVB after a remote p2p peer leaves, the remote tracks (of the peer that just left) are removed from the conference after the SSRCs are removed from SDP and since it necessitates a renegotiation, the task is pushed to the modification queue. Since the switch to jvb connection happens immediately, the remote jvb remote tracks are present after the switch and they get added to the conference again.
Add the check for the remote participant before adding the tracks. This fixes an issue where the remote tracks are present in redux even after the participant leaves.

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.

* 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

* 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'

* ref: Inline onMucMemberLeft (has nothing to do with "moderator" and is broken).

* fix: Use isFocus instead of just reading the resource.

* ref: Inline onParticipantLeft.

* ref: Add _ to private function names.

* ref: Remove redundant function.

* ref: Initialize focusComponent once.

* ref: Remove two copies of "options".

* ref: Simplify the focusJid flow.

* ref: Rename focusCompoonent to targetJid.

* fix: Only retry on "invalid session" errors

We used to fire FOCUS_DISCONNECTED (which doesn't have anything to do
with anything disconnecting) and then retrying. This is not necessary
because FOCUS_DISCONNECTED is a fatal error and the client schedules
a reload of it's own (and we want the client reload as opposed to the
conference-request retry in case e.g. the shard has changed).

* ref: Remove use of visitorFocus.

* ref: Remove graceful shutdown handling (never used in backend).

* feat: Support conference request over HTTP.

* ref: eslint cleanup.

* fix: Add max timeout 2 minutes.

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.

(Rather than moving streams to inactive in SDP.)

* fix: Do not send empty <nick/> when a nick exists and gets removed

* squash: Fix lint errors.

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.

Workarounds around issue #2037 to have other participants receive the mute event and therefore avoid frozen camera bug

It implements SAS verification as per the Matrix spec, adapted to our environment.

* feat(face-landmarks): add timeline

* add duration to xmpp payload

* 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.

fix: corrected error message (see 6810839edc)

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>

This fixes an issue where mute camera operation doesn't stop sending camera stream even though locally it appears to the user that they are muted. This happens when multiple camera streams are added to peerconnection because of how toggle of the video button is implemented. This limitation will be removed when the application is fixed.

Avoid needing an external dependency.

Allows a moderator to end the conference for everyone.

Co-authored-by: Werner Fleischer <70745309+wernf@users.noreply.github.com>

Co-authored-by: Дамян Минков <damencho@jitsi.org>

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.

In most cases it's not a real issue and it's still part of the logs,
throwing an unhandled exception at the global handler just prints an
ugly trackeback in the JS console which looks much worse than it really
is.

* fix(multi-stream): Fix local SSRC cache to include multiple video streams.
If multiple local video streams are found in the SDP, cache all of them instead of the first video SSRC. This fixes an issue where the resolution/fps stats for the screenshare track are not available.

* squash: new track inherits the source name of old track if it exists.

Removing and adding a track back to the conference doesn't generate new source signaling all the time so presence needs to be updated every single time.
Fixes https://community.jitsi.org/t/screen-share-doesnt-work-a-second-time-on-p2p/114024 and possibly https://github.com/jitsi/lib-jitsi-meet/issues/1997.

With just source-name enabled, set the source name of the replaced track before configuring the encodings, this fixes an issues where the sender constraints are not applied on the p2p connection because the source name is undefined.

* fix(video-quality) Update frame heights in content-modify for p2p.
When the user changes the preferred video quality settings from the UI, update the frame height values in content-modify for p2p connection when source-name signaling is enabled.

* fix(multi-stream) Send presence for desktop track before signaling when it is the first track to be added to the conference.
Also do not suppress renegotiation for desktop track on p2p since the browser doesn't fire negotiation needed event.

* squash: fix build

* squash: fix linter issues

* squash: add comment.