Since Track effects and mute/unmute operations use the same flow, i.e., removing/adding the track from the RTCPeerConnection but not from TPC, make the names of the methods involved more generic.

Since the track is only temporarily removed and added back, it can be treated like mute & unmute operation. Fixes https://github.com/jitsi/lib-jitsi-meet/issues/2048. Remove a TODO thats no longer needed, track replace operation don't force renegotiations anymore.

They are always required and the signaled features are ignored.

* feat: Add videoType to sources signaled in jingle.

* Add a receiveMultipleVideoStreams flag.

The identities are coming from the presence and all values there are string.

It's not necessary to perform a full join, sending a conference IQ is
enough.

Set the bitrates on the video m-lines based on the associated track type. In multi-stream mode, there will be two separate m-lines for camera and desktop tracks. Also fallback to SDP munging for setting codec preferences on Chromium.

The hack for making Chrome capture the highest resolution possible for desktop track breaks high fps screen capture.

* fix(SDP): don't separate fmtp parameter with spaces

* adjusted tests accordingly to not expect spaces anymore

Reusing(removing it and adding it again) an SSRC in the
remote SDP is leading to the issue where remote video is received but
not rendered. That's why we shouldn't remove the receive only SSRCs
from the SDP. This way the sender won't send any source-remove and
source-add for the same SSRCs.

the editor will no longer complain about superfluous await when calling

it removed support for functions being simultaneous async and have done parameter in callback

the editor will no longer complain about superfluous await when calling these functions

* fix(SignalingLayer): Skip checking for last presence if the client doesn't join muc.
This fixes the broken wireless screensharing issue on spot clients.

* squash: change log message to a warning.

Jicofo by default now strips the simulcast SSRCs and sends only the primary SSRC and RTX to all the receiving endpoints in the call. Therefore remote SDP munging is not required on the clients anymore.

see https://community.jitsi.org/t/lib-jitsi-meet-jwt-support-with-websockets/79806

In most cases it's not a real issue and it's still part of the logs,
throwing an unhandled exception at the global handler just prints an
ugly trackeback in the JS console which looks much worse than it really
is.

related to 59665767bf

* added docs on how to define tokens

* fixed typo

Co-authored-by: James Green <james.green@affinity-digtal.com>

* fix(deployment-info) Prevent error when setting shard and region

* fix

They have been deprecated and their valued can no longer be changed from
the default after M103.

Ref: https://groups.google.com/u/0/g/discuss-webrtc/c/2l7jWgreYpw?pli=1

This reverts commit 7d3f13c907.
This change was not needed.

RN clients throw a function undefined error otherwise.

Adds it to deploymentInfo.

When source-name signaling is enabbled on plan-b clients, the sources to MSID map needs to be cleared after every transformation since browser in plan-b mode produces a new set of SSRCs and trackID when a track is removed and new track is added to the peerconnection. This fixes an issue where the track index in the source-name gets incremented after a track replace operation.

* fix(multi-stream): Fix local SSRC cache to include multiple video streams.
If multiple local video streams are found in the SDP, cache all of them instead of the first video SSRC. This fixes an issue where the resolution/fps stats for the screenshare track are not available.

* squash: new track inherits the source name of old track if it exists.

* fix(rtc): fix local track removeTrack promise conference check

* ref: ignore if promise was successful or not

Enables lite mode which disables any media decoding.

This allows us to display resolution and framerate stats on separate tiles when multi-stream mode is enabled.

Removing and adding a track back to the conference doesn't generate new source signaling all the time so presence needs to be updated every single time.
Fixes https://community.jitsi.org/t/screen-share-doesnt-work-a-second-time-on-p2p/114024 and possibly https://github.com/jitsi/lib-jitsi-meet/issues/1997.

With just source-name enabled, set the source name of the replaced track before configuring the encodings, this fixes an issues where the sender constraints are not applied on the p2p connection because the source name is undefined.

With that, it's possible to set AV1 as preferred codec for p2p and try it on Chrome.