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.

This fixes a regression introduced by the previous commit.

sLD is called immediately after createOffer/createAnswer, therefore the desc provided by createOffer/createAnswer can be directly passed to sLD without the need for converting it to unified plan format. This also fixes a warning seen on the browser console that says 'The description does not look like plan-b'.

The issue currently affects browsers based on Chromium 96.0.4664.110
including Brave, Chrome, Edge & Opera and Google Chrome 97.0.4692.56

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

* squash: Add todo comment.

* feat: Adjust setReceiverConstraints to use new format

* add ReceiveVideoController spec

* rename prioritizedSources to onStageSources

* call FeatureFlags.init in specs and update description

* provide a default flags argument to FeatureFlags.init

* fix(RTC): fixed typo in function name

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

* fixed typos

* fix(tpc) disable ulpfec on chrome 96

* ref(tpc) add comment

Broken with previous commit 579a49.

* 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.

This fixes an issue where a user is not able to unmute their video if the MediaStreamTrack associated with the camera stream returns a null value for the track height.

* 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>

This ensures that the source signaling is sent before the mute state is sent in presence. Jicofo relies on mute state from presence to check if the sender limit has been reached.

* 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

Sends the source name to video type mapping to the jvb
in a new bridge channel message.

* fix(sdp): provide SCTP streams, because the XMPP parser expects them

* fixed linting error

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

This PR switches to generating sctp-port based SDP.
It also adds support for parsing sctp-port, but can handle
fallback to old-style sctpmap as well.
Fixes: #1778

the protocol was switched back in Chrome 44 in 2016:
  https://bugs.chromium.org/p/webrtc/issues/detail?id=2796
the port switch more than five years ago too.

Add timestamp to all tracks, and display surface to screen sharing tracks

This fixes occasional failures of MuteTest.MuteAfterJoinCanShareAndUnmute torture test and also the case on Safari where user stopping the screenshare doesn't stop showing the screensharing indication on the thumbnail.

...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.