Since the order of the ssrcs in the json-encoded message is not guaranteed to be in the correct SIM/FID order, log all the ssrcs.

* ref(JingleSession) Recycle m-lines by rejecting them on source removal.
By rejecting the m-lines on source removal (setting the port to 0 on the associated m-line), m-lines can be recycled for new sources instead of appending new ones.

* chore(deps) update sdp-interop@latest.

* fix: Reads lobby jid from correct location in the error.

* squash: Add todo comment.

* fix(RTC): fixed typo in function name

* fix(xmpp): codec renegotiation task needs to be queued

* fixed typos

* feat: Adds events for session-add, session-remove and session-accept.

* feat: Refactor updating presence for audio/video mute and video type.

This brings few changes and fixes. Thew initial presence will always miss audioMuted and videoMuted value, those will be added on session accept. All updates to presence are done on sessionAccept, on source-add or on source-remove and the only one not signalled when camera track is replaced by video track and vice versa.
This change brings more presence updates when number of participants are below startAudioMuted/startVideoMuted, but when participants are above those numbers we get less presence update. This is important for big meetings.
Fixes wrong videoMuted state, as replace track and mute are both executed in promise, sometimes the replace-track one finishes first and when the mute one is resolved there is no conference object in the track to be able to update the presence (hitting this when we pass the startAudioMuted threshold).

* squash: Put back the tracks for _setTrackMuteStatus and _setNewVideoType.

* squash: Fix line length.

* squash: Fix skip sending presence twice.

* squash: Adds the error to the error callback.

* squash: Fix newJingleErrorHandler.

Promise.all will be rejected with the first rejected Promise, without waiting
for the rest, we don't want that.

If leaving a room results in a timeout, we wouldn't be cleaning up the EMUC
state, which will prevent us from joining the room again, as it would detect it
as a duplicate.

Both of this features listen for events in the XMPP connection. Thus, if we join
another meeting over the same connection we might still receive some events on
the old event handler.

Stop listening for events from the components as soon as the leave operation is
triggered.

Jicofo sends a presence when the audio/video sender limit is reaced in the conference. The client can then proceed to disable the audio and video mute buttons when this occurs.

This is necessary when adding new properties to the payload.

* Save track source name to JitsiRemoteTrack

* add feature flag checks. code review adjustments.

* eslint cleanup1

* eslint cleanup2

* fix: move sourceName declaration outside of if block.

Co-authored-by: Dennis Dowhy <ddowhy@gmail.com>
Co-authored-by: William Liang <wliang67@bloomberg.net>

They are companion rooms created in a separate MUC. The room relationship is
maintained by a Prosody plugin.

All signalling happens through the breakout rooms MUC component.
Co-authored-by: Saúl Ibarra Corretgé <saghul@jitsi.org>

* feat: facial expression in speaker stats

* feat: send xmpp message with facial expression to server

* fix: rebase conflicts

* feat: facial expression in speaker stats

* feat: facial expression in speaker stats

* fchore(facial-expressions): remove the send facial expression call from update facial expression function

* feat(facial-expressions): store expresions as a timeline

* refactor(facial-expressions): store expressions by counting them in a map

* feat(facial-expressions): camera time tracker

* add(facial-expression): increase facial expression with duration parameter from payload

* add(facial-expressions): the disgusted expression

* refactor(facial-expressions): remove camera time tracker

* refactor(facial-expressions): change data channel message handel position for facial expressions and renamed some types

* fix(facial-expressions): move facial expression endpoint message handler from statistics.js to ConnectionQuality.js

* fix(facial-expressions): remove unused type

RTX in combination with simulcast on Firefox results in bad video quality
therefore we only enable it for version >= 96.

...to advertise track's muted state and the video type.

For now, if the source name signaling flag is enabled, both legacy and
new <SourceInfo> element will be used at the same time. This is to be
able to interoperate with legacy clients and start testing the new
format at the same time. Whenever possible <SourceInfo> will be used
as main source of truth with the fallback to legacy <audiomuted/>,
<videomuted/> and <videotype/> elements.

* feat: Delays deployment info stats till we get update from backend.

We delay sending this stat till we are connected or fail to connect.

* squash: Make sure we send it on conn fail or when sys message or disco info is received.

Fixes https://github.com/jitsi/jitsi-meet/issues/9988.

We do not have to wait for the removal of the ssrcs from the remote description for removing the remote tracks associated with a participant that left the call. This speeds up removal of the participant from call even if the JingleSession modification queue is backed up.

This fixes https://github.com/jitsi/lib-jitsi-meet/issues/1759.

Instead rely on the 'negotiationneeded' event fired by the browser for JVB connection. This makes local source changes faster even if the modification queue is backed up.

* feat: Adds to JitsiConferenceEvents CONFERENCE_JOIN_IN_PROGRESS.

* squash: Fixes tests.

* ref(TPC) Configure encodings in one go.
Configure the active state, maxBitrate and degradation preference in one go instead of configuring them separately.

* squash: Address review comments.

* squash: fix a comment.

This is the first step in adding support for multiple
streams per endpoint. Each source(stream?) needs to have
an identifier. For now always generate the 0 index name
until the machinery for sending more than 1 stream is put
in place.

* send only MSID attribute

* add feature flag for source name signaling

* log a msg if source name signaling is enabled

Maybe due to our DOM wrappers, the Symbol.iterator is not implemented, so
iterate over the IQ fields old-school style.

If a second JitsiConference instance for the same room is created we were
throwing an error but some listeners were already attached. This commit
makes sure that we throw the error as soon as possible and no listeners
are added.

ChatRoom.setAudioMute and ChatRoom.setVideoMute are
not async, so no callbacks nor promises are necessary.

Instead of logging the full IQs for Jingle messages like session-initiate, source-add and source-remove which can be very long, log just the formatted source information.

It makes more sense to call it pcConfig since it is the RTCConfiguration object passed to the WebRTC peerconnection.

* feat(av-moderation) Remove from moderation whitelist functionality

* Fix comment

Fixes issue with jibri that are served on different virtual host and jiconop2 does not work for them.

We already log if a focus leaves, which the the important bit. This avoids a log
flood when leaving a large conference.

* feat: Support plain endpoint ID.

Support the ID of the owner of sources being encoded directly instead of
using the full JID.

* feat: Support JSON-encoded sources.

* feat: Advertise support for JSON-encoded sources.

* fix: Updates shard value in XmppConnection on new value from xmpp.

* squash: Start ws keep alive only after the successful connection.