* feat(rtcstats): Adds a default log collector.

* squash: Fix comments.

* squash: Fix comments2.

* squash: Fix comments.

Aparently we call this for all tracks including for the audio tracks and it is failing with JS error. This leads to "participant unable to unmute".

* fix(SDP) Add missing trackID if its missing in msid.
This is needed for older versions of Chrome. Also skip performance optimizations if encode resolution is missing in the stats.

* squash: address review comments

Wrong check for unsupported getDisplayMedia when starting screen sharing will cause the app to never fallback to getUserMedia flow. This breaks screen sharing for all old electron clients that do not support getDisplayMedia.

Reconstruct the msid to include both streamId and trackId if its missing from the source signaling. Fixes issue on Chromium < 117 where the remote offer gets rejected and the client gets kicked out of the call.

Firefox added support for AV1 with DD and VP9 SVC is also working as expected in Firefox 136.

Detect if gDM support has been enabled and fallback to the previous gUM
flow otherwise.

- Lower logger level to warning
- Use nested conditions

If the enclosing Jitsi Meet app declares gDM in Electron is supported,
ignore the `electronUseGetDisplayMedia` config flag and use the gDM
flow. This facilitates having backwards compatibility since the Electron
app won't know if the deployment it's about to join has the gDM flow
support or not.

This should be supported by the backend. This limitation was added here only because of track synchronization issues between JM and LJM. The support is added behind testing.allowMultipleTracks config.js setting
Fixes https://github.com/jitsi/lib-jitsi-meet/issues/2205

* feat: Updates userRegion from backend.

* squash: Update comment.

* squash: Fix options.

* fix(codec) Debounce the call that calc codec intersection set.
Calculate codec intersection set only once per second even when there is a burst of joins/leaves. Also, check for current codec before chaining a codec change operation when codec selection API is used.

* squash: Address review comments

Jigasi / Jitsi (desktop) seems to encode apostrophes like (&apos or &#39) for
instance.

This is notably visible in the subtitles.

Normalize it so it's plain UTF-8.

* fix(SDP) Include the trackId in the signaled msid for the source.
Without the trackID in a=ssrc lines in SDP, older versions of Chrome (108) reject the remote description resulting in endpoint getting kicked out of the conference.

* squash: Fix unit test

Avoids having any actual remote audio tracks, which can mess with the
mobile audio session.

The calls to RTCRtpSender.setParameters() are all chained and the current maxHeight is set after the call to setParameters is resolved. If there is another call made to setParameters before the previous one resolves, we can end up passing the wrong maxHeight resulting in the client getting stuck at an unexpected resolution. This issue can be reproduced sometimes  when the users are moving across the main and breakout rooms. TPC.setVideoCodecs() ends up pushing a wrong maxHeight for update when the previous call to setParameters hasn't resolved yet.

* feat(subject): Avoids setting subject if already set.

* squash: fix comments.

* squash: fix comments.

Don't show the misleading "user cancelled" error, there are corner cases
in which a different error can be manifested.

Fixes a regression introduced by a7476b126c where multiple source-adds are sent from Firefox client resulting in Jicofo force muting the sources.

* feat(RTC) drop PERMISSION_PROMPT_IS_SHOWN event

It was never accurate since browsers have changed their behavior over
time and can be implemented in the application more reliably.

* squash: Update JitsiMeetJS.ts

* squash: Update hand-crafted JitsiMeetJS.d.ts

---------

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

Some so called "security" browser extensions override WebRTC APIs so
even if the UA string suggests the browser is supported, it won't work.

Coincidentally, this should also mark Safari in Lockdown Mode as
unsupported, since the WebRTC APIs are not available in that case.

Regression was introduced in 07f037e12e/. Also process encodeTime stats when H.264 is selected

The constructor for RTCSessionDescription has been deprecated - https://developer.mozilla.org/en-US/docs/Web/API/RTCSessionDescription/RTCSessionDescription.

* fix(TPC) Use videoType from 'source-add' for remote track creation.
If 'source-add' for a remote video source is received before presence for that source, videoType will default to 'camera' and the client wouldn't be able to create the virtual participant tile for rendering the desktop track.

* squash: Include the videoType for no SSRC-rewriting case.

Fixes a regression for JitsiConference::setReceiverVideoConstraint and JitsiConference::setAssumedBandwidthBps.

If guest authenticates in meeting to become a moderator and creates a breakout room, was not inviting jicofo and fails to join breakout room.
Fixes jitsi/jitsi-meet#15024.

* feat(JingleSession) Convert Jingle->SDP directly w/o interop layer.

* Update modules/RTC/TraceablePeerConnection.js

Co-authored-by: Saúl Ibarra Corretgé <s@saghul.net>

* ref(SDP) Introduce an enum for ssrc-group semantics.
Address review comments, fix test faiilures for ssrc-rewriting case and add unit tests.

* Update modules/RTC/TraceablePeerConnection.js

Co-authored-by: Saúl Ibarra Corretgé <s@saghul.net>

* squash: Address review comments

---------

Co-authored-by: Saúl Ibarra Corretgé <s@saghul.net>