...to advertise track's muted state and the video type.

For now, if the source name signaling flag is enabled, both legacy and
new <SourceInfo> element will be used at the same time. This is to be
able to interoperate with legacy clients and start testing the new
format at the same time. Whenever possible <SourceInfo> will be used
as main source of truth with the fallback to legacy <audiomuted/>,
<videomuted/> and <videotype/> elements.

* feat: Delays deployment info stats till we get update from backend.

We delay sending this stat till we are connected or fail to connect.

* squash: Make sure we send it on conn fail or when sys message or disco info is received.

Fixes https://github.com/jitsi/jitsi-meet/issues/9988.

We do not have to wait for the removal of the ssrcs from the remote description for removing the remote tracks associated with a participant that left the call. This speeds up removal of the participant from call even if the JingleSession modification queue is backed up.

This fixes https://github.com/jitsi/lib-jitsi-meet/issues/1759.

Instead rely on the 'negotiationneeded' event fired by the browser for JVB connection. This makes local source changes faster even if the modification queue is backed up.

* feat: Adds to JitsiConferenceEvents CONFERENCE_JOIN_IN_PROGRESS.

* squash: Fixes tests.

* ref(TPC) Configure encodings in one go.
Configure the active state, maxBitrate and degradation preference in one go instead of configuring them separately.

* squash: Address review comments.

* squash: fix a comment.

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.

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.

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

* Fix comment

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

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

* 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.

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

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

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

The prosody toJson returns for empty arrays {} which causes an error initially.

Hurts mobile performance. They can still be enabled by setting the level at the
right value (debug / trace).

Hurts mobile performance. They can still be enabled by setting the level at the
right value (debug / trace).

Chrome doesn't create a decoder for ssrc in the remote description when there is no local source and the endpoint is offerer. Initiating a renegotiation with the endpoint as a responder fixes this issue. Add a workaround until Chrome fixes this bug.

Do not enable unified plan for p2p chrome by default until StartMutedTest is fixed.
Fix media direction for case when there are no local and remote sources, should be set to 'inactive' in that case.

In unified plan, re-use of m-line (i.e., adding an SSRC, removing it and then adding it back) causes the browser to not render the media on Chrome and Safari. The WebRTC spec is not clear as to how browsers have to behave, this doesn't cause any issues on Firefox. As a workaround, only change the media direction and leave the ssrc in the remote desc. This automatically triggers a 'removetrack' event on the associated MediaStream and the track can be removed from the UI.

For p2p connections, the media direction needs to be adjusted after every source-add/source-remove is processed based on the availability of local sources.

Disable cross browser p2p using unified plan until all the issues are fixed.

The 'signalingstatechange' event for 'stable' is fired after the 'iceconnectionstatechange' event for 'completed' is fired on chrome in Unified plan. This prevents the client from switching the media connection to the p2p connection once the ice connection for p2p gets established.

Add a preifx to logs for idenitifying the type of TPC/jingleSessionPC.

This can be controlled through the config.js  option 'enableUnifiedOnChrome'.

* fix(xmpp): use text parameter instead of attr for message

bug: msg with elementName != 'body' ->
   Cannot read property 'appendChild' of null

fix leads to another problem - there is some magic in the 'c' function:
"This function moves the current element pointer to the child,
unless text is provided"

apply the same magic here. will be removed in the next commit

* fix(xmpp): do not rely on internals of Strophe.c

* squash: Fixes tests.

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

The direction was marked as 'inactive' only on Firefox as Safari had audio issues when an inactive mid is re-used. Chrome (in unified-plan) needs the direction of the mid in remote desc to be set to 'inactive' for a 'removetrack' to be fired on the associated media stream whenever a remote source is removed.

If the msid attribute is missing, then remove the ssrc from the transformed description so that a source-remove is signaled to Jicofo. This happens when the direction of the transceiver (or m-line) is set to 'inactive' or 'recvonly' on Firefox. Not signaling these source updates creates issues with remote track handling on the other endpoints in the call.

There is a known issue with Firefox where the BWE gets halved on every renegotiation causing the low upload bitrates from the Firefox clients.

Fixes https://github.com/jitsi/docker-jitsi-meet/issues/1048

* feat: Audio/Video moderation.

* squash: Fix docs.

* squash: Adds some warning logs when execution is rejected.

* squash: Changes a field name in the message for adding jid to whitelist.

* squash: Send to participants only message about approval.

Skips sending the whole list.

* squash: Fixes tests.

* squash: Adds more logs.

* feat: Separates enable/disable by media type.

Adds actor to the messages to inform who enabled it.

* squash: Fixes log line.

* squash: Fixes comments.

* squash: Fixes log messages.

* squash: Fixes comments.