When the participant has joined after A/V moderation has been enabled
they are auto muted with the startmuted node in the session-initiate
packet. We were not marking those as muted by the focus and therefore on
unmute we were not sending the <muted>false</muted> IQ to jicofo.
The result was that the bridge wasn't forwarding the media to the
remote participants.

When an answer is received from the p2p peer, only set remote description instead of calling the full renegotiation cycle. Also, always initiate a sRD->cA->sLD cycle when adding remote sources irrespective of whether the local endpoint is an initiator or responder for the p2p connection. This lets us remove the hacks added for Chromium (unified-plan) bugs related to video not rendering in p2p cases.

fix up test failure by including extends Listenable

* modules: add RFC 2198 audio redundancy encoder

which can be used together with WebRTC Insertable Streams to encode
a variable amount of audio redundancy

* port some tests

* port the rest of the tests but the comparison seems broken

* fix most lint

* with jasmine@4 actually fix the tests

* fix lint issues

* +jsdoc

* +nils review

and update other test dependencies

It does no harm on RN, since it doesn't detect it as a supported browser
and bails out.

It gets resolved during translation.

Fixes: https://github.com/jitsi/lib-jitsi-meet/issues/2055

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>