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.)

<conference xmlns="http://jitsi.org/protocol/focus" vnode="v1" ready="true" room="roomtest@conference.focus.exmaple.com" focusjid="focus@auth.meet.jitsi">
    <property name="authentication" value="false"/>
    <property name="sipGatewayEnabled" value="true"/>
</conference>

* Remove callstats conference property.

* Remove RTX conference property (removed from backend).

* Remove audioPacketDelay conference property (removed from backend).

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.

Delete hasAnyVideoTrackWebRTCMuted method as it is not used anymore.

* 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.

It's now in JitsiMeetJS.ts rather than .js.

<iq id='some-id' from='focus.exmaple.com' type='error' xml:lang='en-US' to='some-jid' xmlns='jabber:client'>
    <error type='modify'>
        <redirect xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/>
        <conference xmlns='http://jitsi.org/protocol/focus' vnode='v1' main-room='roomtest@conference.focus.exmaple.com' focusjid='focus@auth.meet.jitsi'/>
    </error>
</iq>

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

It's emitted when remotely closed / failed.

Bumps [loader-utils](https://github.com/webpack/loader-utils) from 1.4.0 to 1.4.2.
- [Release notes](https://github.com/webpack/loader-utils/releases)
- [Changelog](https://github.com/webpack/loader-utils/blob/v1.4.2/CHANGELOG.md)
- [Commits](https://github.com/webpack/loader-utils/compare/v1.4.0...v1.4.2)

---
updated-dependencies:
- dependency-name: loader-utils
  dependency-type: indirect
...

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

* 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.

The candidates are not necessary when we use jvb.

Use the participant ID as the key to codecs, etc, instead of the source
name. The object is further indexed by SSRC.

Don't wait for both codecs to be set before propagating codec
information.

Sample data model for codecs:

codec: {
    participant1: {
        ssrc1: {
            audio: 'opus',
            video: undefined
        },
        ssrc2: {
            audio: undefined,
            video: 'VP8'
        },
        ...
    }
    ,...
}

Some values, like codecs, are available right away, we don't need to
compute differences.

fix: corrected error message (see 6810839edc)

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.