We need to make sure the audio track is added to the JVB connection, or we won't
be able to unmute.

Why this happens is a mystery wrapped in an enigma, it started happening with
iOS 15.

Same thing applies to Safari for macOS.

Fixes: https://github.com/jitsi/jitsi-meet/issues/10104

This is the first step in adding support for multiple
streams per endpoint. Each source(stream?) needs to have
an identifier. For now always generate the 0 index name
until the machinery for sending more than 1 stream is put
in place.

* send only MSID attribute

* add feature flag for source name signaling

* log a msg if source name signaling is enabled

Maybe due to our DOM wrappers, the Symbol.iterator is not implemented, so
iterate over the IQ fields old-school style.

There was an error in the pattern match, everything was included in it. In
addition return a boolean in case we succeed in the UA pattern match, since it's
more correct than returning an array with the match.

If a second JitsiConference instance for the same room is created we were
throwing an error but some listeners were already attached. This commit
makes sure that we throw the error as soon as possible and no listeners
are added.

ChatRoom.setAudioMute and ChatRoom.setVideoMute are
not async, so no callbacks nor promises are necessary.

Instead of logging the full IQs for Jingle messages like session-initiate, source-add and source-remove which can be very long, log just the formatted source information.

It makes more sense to call it pcConfig since it is the RTCConfiguration object passed to the WebRTC peerconnection.

Log only the important events that we care about on the HTMLMediaElement that the remote tracks are attached to.

* feat(av-moderation) Remove from moderation whitelist functionality

* Fix comment

Credit to Stephen Donner.

For some reason webpack seems to not be transpiling that properly.

Fixes: https://github.com/jitsi/lib-jitsi-meet/issues/1722

This fixes an issue where the bridge doesn't forward the HD stream from Firefox to other users in the call. The order of the ssrcs produced by the browser is from Highest resolution to lowest whereas the bridge assumes it to be from lowest to highest as is the case in Chrome and Safari.

If Jicofo doesn't offer VP9 but the client expresses a preference for VP9, VP9 bitrates were being imposed before.

Fixes issue with jibri that are served on different virtual host and jiconop2 does not work for them.

This swallows the stack trace.

If oldTrack was not previously in the conference this will
lead to a crash in onLocalTrackRemoved where oldTrack doesn't
have `muteHandler` and `audioLevelHandler` listeners defined.

This is needed for screensharing to work as expected on chrome in unified plan.

... is stopped. If there's any sRD/sLD cycle happening
while the screen sharing stops, the local track
returns muted and it was injecting fake video SSRCs
which results in invalid SSRC description.

At the time when it happens there's error log in jicofo:
"Error adding SSRCs from X"
and the client will get bad-request error in response
to the source-add request.

There is no need to check if we need to sswitch from P2P to JVB or vice versa if
we have already left the room. It creates a log flood.

We already log if a focus leaves, which the the important bit. This avoids a log
flood when leaving a large conference.

On mobile Safari, if a user joins audio and video muted, the browser doesn't decode the incoming audio. Workaround is to always add the audio track to pc and mute it if needed.

* feat: Support plain endpoint ID.

Support the ID of the owner of sources being encoded directly instead of
using the full JID.

* feat: Support JSON-encoded sources.

* feat: Advertise support for JSON-encoded sources.

Fixes error when trying to stop screen sharing and video moderation is on

* fix: Updates shard value in XmppConnection on new value from xmpp.

* squash: Start ws keep alive only after the successful connection.

An alternative to Insertable Streams, currently implemented in Safarii / WebKit.

https://w3c.github.io/webrtc-encoded-transform/

Fixes: https://github.com/jitsi/jitsi-meet/issues/9585

If audio levels are disabled it's pointless to try and detect problems.

* fix: re-enable that ICE connection failures in Chrome are detected
* fixed linting errors

* enable config callstats configParams

* add comments

* wrap long comments

* fix format

* fix format

* fix format

* add jsDoc