* feat(codec-selection): Use the new codec selection API
https://github.com/w3ctag/design-reviews/issues/836. This allows the client to seamlessly switch between the codecs without having to trigger a renegotiation.
This feature is behind the flag testing.enableCodecSelectionAPI in config.js

* fix(stats): Fix local resolution stats.
The video codec for the local video sources needs to identified differently now, from the codecs field in the RTCRtpSendParameters returned by the browser. We no longer munge the remote SDP to switch to a different codec.

* feat(stats): Feed encodeTime stats for all local SSRCs to the codec selection mechanism.

* fix(codec-selection) Continue to mumge SDP for selecting H.264.

* feat(codec-selection) Make screenshare codec configurable.
If no 'screenshareCodec' is set under videoQuality or p2p settings, AV1 will be selected as default.

* squash: Address review comments

* Update modules/RTC/CodecSelection.js

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

* fix(codec-selection) Add codec to existing stats

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

Use an exponential backoff time for initiating ICE restart to prevent loading the prosody.

When a large number of endpoints loose connection to the signaling and media servers because of an ISP failure, these endpoints will try to reconnect at the same time causing a storm of source-remove/source-add events to the signaling servers that can lead to prosody going down otherwise.

When an existing SSRC for a remote track gets re-mapped from one source to another, fire a TRACK_REMOVED event followed by TRACK_ADDED event instead of TRACK_OWNER_CHANGED event. This should simplify the application logic for track handling.

Also fixes the issue with video freezing after video mute and unmute when the user joins the call with startSilent=true.

This reverts commit 006457faa0.

Include visitor codecs published by Jicofo while calculating the intersection set for the conference.

The tracks should get removed if the startAudioMuted/startVideoMuted flags are set for the conference. Fixes the failing startMuted testcase.

This makes sure that any track operations that are executed after the terminate is sent and before the new session is established get synced up on the new session.
Also reset the JVB stats. Fixes https://github.com/jitsi/jitsi-meet/issues/14326.

Add debug info.

When tracks are not included in the initial offer/answer, remove them from the conference. Otherwise, the tracks in conference will be out of sync with those in the media sessions.

If the local track was not added to the conference because the participant was muted on join by focus, add it the conference instead of adding it to the media session when the user unmutes. Fixes a failing torture test case.

* fix(RTC): Remove support for plan-b SDP.
Since unified plan mode becomes the default mode, so does multi-stream mode.

* squash: Remove plan-b unit tests and fix comments

* squash: address review comments

When visitors are joining or leaving (from 0 to 1 or just going to 0) we need to stop/start p2p.

Fixes an issue with audio after p2p->jvb switch when the media active state is set to false because of a setVideoCodecs call after a remote user joins the call.

* fix(events): unify events to a single EventManager type, add support for single typescript declaration

* fix(lint): fix lint

* fix(events): fix incorrect instatiation

* fix(events): clean up redundant methods

* fix(events): keep EventEmitter name, alias NodeEventEmitter

* fix(events): fix loose reference

* fix(EventEmitter): remove on/off alias as redundant

* fix(RTCUtils): bring event handlers under class to use same event emitter

* fix(RTCUtils): fix lint

Since there is nothing to do, pretend it all went well.

* ref: Remove PrecallTest.

* ref: Remove callstats support.

There is no reason to wait for for the event to be emitted.

It can be implemented at the application layer if desired.

* Added rtcstats

* increased webpack performance size warning to 830K

* Addressed most of Saghuls and Andrei's review comments.

* moved rtcstatsUseLibJitsi from analytics to testing

* refactor ljm rtcstats, use ljm conference events

* remove debug logs

* remove wrongly named files

* rename rtcstats module

* fix test

* address code review

* fix pack error

* address code review

---------

Co-authored-by: Nils Ohlmeier <github@ohlmeier.org>

* fix(codec-selection) Use a configurable preferred codecs list to select the codecs.
Allow asymmetric codecs to be configured on the endpoints. This means that Firefox and Safari which have bugs with VP9 encode will now encode VP8 but decode VP9 coming in from Chromium endpoints.

* fix(codec-selection) Add unit tests for the codec selection logic.

* feat(codec-selection): Introduce mobileCodecPreferenceOrder setting in config.js.

* fix(codec-selection) Avoid unnecessary renegotiations.
On participant join/leave, check if the new intersection of codecs are already configured to be the top n codecs.

* Address review comments

* fix: Strip the codecs that are not in the codec list from the video m-lines.
Also, ignore remote codecs published in presence for p2p connections.

* fix: Define default codec order for mobile and desktop.
Fix an issue where p2p between mobile and desktop was broken.

* fix: Add default codecs for both p2p and jvb on mobile devices.
Munge the initial offer sent out by RN clients since RN doesn't support RTCRtpTransceiver#setCodecPreferences.

* feat(signaling) Only include 'msid' in the source description.
'cname', 'label' and 'mslabel' attributes that are included in the source description of the signaling messages are ignored by Jicofo. Only 'msid' is used for source validation.
Therefore, drop those attributes and make sure 'msid' is always included and in the format '<endpoint_id>-<mediaType>-<trackIndex>-<tpcId>'.

* fix(p2p) Enable p2p for Firefox.
Set the senders attribute correctly in the outgoing jingle message. Ignore the media direction from the session, only look for the direction in the media desccription. This fixes an issue where Firefox rejects an answer because the direction is not set correctly.

Fixes an issue that started surfacing after f0dd403951 because local tracks were getting added to pc even when the moderator has set start muted policy to true.

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.