* feat(multi-stream) Enable multi-stream by default (if not disabled explicitly).

It's only used for the UMD bundle.

It does the minimum necessary to function without the complexity of the
main one.

Fixes cases where SS stream is not being sent when p2p->jvb switch happens.

Allows a moderator to end the conference for everyone.

Co-authored-by: Werner Fleischer <70745309+wernf@users.noreply.github.com>

This fixes an issue where the bitrates for screenshare were much higher than before for VP9 causing the JVB to suspend SS streams more often.

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

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

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

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.

* feat: Add videoType to sources signaled in jingle.

* Add a receiveMultipleVideoStreams flag.

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 editor will no longer complain about superfluous await when calling

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.

related to 59665767bf

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.

Enables lite mode which disables any media decoding.

* fix(video-quality) Update frame heights in content-modify for p2p.
When the user changes the preferred video quality settings from the UI, update the frame height values in content-modify for p2p connection when source-name signaling is enabled.

* fix(multi-stream) Send presence for desktop track before signaling when it is the first track to be added to the conference.
Also do not suppress renegotiation for desktop track on p2p since the browser doesn't fire negotiation needed event.

* squash: fix build

* squash: fix linter issues

* squash: add comment.

This fixes an issue where the receiver constraints and video type messages are sent on the bridge channel only the first time the ws conn gets re-established.