Fixes an issue with audio after p2p->jvb switch when the media active state is set to false because of a setVideoCodecs call after a remote user joins the call.

* fix(events): unify events to a single EventManager type, add support for single typescript declaration

* fix(lint): fix lint

* fix(events): fix incorrect instatiation

* fix(events): clean up redundant methods

* fix(events): keep EventEmitter name, alias NodeEventEmitter

* fix(events): fix loose reference

* fix(EventEmitter): remove on/off alias as redundant

* fix(RTCUtils): bring event handlers under class to use same event emitter

* fix(RTCUtils): fix lint

Since there is nothing to do, pretend it all went well.

* ref: Remove PrecallTest.

* ref: Remove callstats support.

There is no reason to wait for for the event to be emitted.

It can be implemented at the application layer if desired.

* Added rtcstats

* increased webpack performance size warning to 830K

* Addressed most of Saghuls and Andrei's review comments.

* moved rtcstatsUseLibJitsi from analytics to testing

* refactor ljm rtcstats, use ljm conference events

* remove debug logs

* remove wrongly named files

* rename rtcstats module

* fix test

* address code review

* fix pack error

* address code review

---------

Co-authored-by: Nils Ohlmeier <github@ohlmeier.org>

* fix(codec-selection) Use a configurable preferred codecs list to select the codecs.
Allow asymmetric codecs to be configured on the endpoints. This means that Firefox and Safari which have bugs with VP9 encode will now encode VP8 but decode VP9 coming in from Chromium endpoints.

* fix(codec-selection) Add unit tests for the codec selection logic.

* feat(codec-selection): Introduce mobileCodecPreferenceOrder setting in config.js.

* fix(codec-selection) Avoid unnecessary renegotiations.
On participant join/leave, check if the new intersection of codecs are already configured to be the top n codecs.

* Address review comments

* fix: Strip the codecs that are not in the codec list from the video m-lines.
Also, ignore remote codecs published in presence for p2p connections.

* fix: Define default codec order for mobile and desktop.
Fix an issue where p2p between mobile and desktop was broken.

* fix: Add default codecs for both p2p and jvb on mobile devices.
Munge the initial offer sent out by RN clients since RN doesn't support RTCRtpTransceiver#setCodecPreferences.

* feat(signaling) Only include 'msid' in the source description.
'cname', 'label' and 'mslabel' attributes that are included in the source description of the signaling messages are ignored by Jicofo. Only 'msid' is used for source validation.
Therefore, drop those attributes and make sure 'msid' is always included and in the format '<endpoint_id>-<mediaType>-<trackIndex>-<tpcId>'.

* fix(p2p) Enable p2p for Firefox.
Set the senders attribute correctly in the outgoing jingle message. Ignore the media direction from the session, only look for the direction in the media desccription. This fixes an issue where Firefox rejects an answer because the direction is not set correctly.

Fixes an issue that started surfacing after f0dd403951 because local tracks were getting added to pc even when the moderator has set start muted policy to true.

Fixes an issue where media gets sent on jvb connection even when the connection is suspended.

Also add all the orphan jvb tracks to the conference when the call switches over from p2p to jvb.

When the call switches over to JVB after a remote p2p peer leaves, the remote tracks (of the peer that just left) are removed from the conference after the SSRCs are removed from SDP and since it necessitates a renegotiation, the task is pushed to the modification queue. Since the switch to jvb connection happens immediately, the remote jvb remote tracks are present after the switch and they get added to the conference again.
Add the check for the remote participant before adding the tracks. This fixes an issue where the remote tracks are present in redux even after the participant leaves.

This fixes an issue where p2p clients (with different codec preferences) fail to decode video because the negotiated codecs are removed from the supported codecs list after the media session is established. The codec preferences will be applied when the first offer/answer is created.

* fix(TPC) Disable media instead of changing dir for p2p->jvb switch.
Resume or suspend the media on the jvb peerconnection by changing the RTCRtpEncodingParamters.active state instead of changing the direction on the transceiver. This avoids the needs to start a O/A renegotiation cycle for these operations. The media direction will be changed only for p2p lastn=0 case since video needs to be disabled on both the sender and the peer for p2p lastn=0 case.

* Address review comments

* Disable media after adding source while media is suspended on the jvb connection. Default 'active' state for stream encodings after the source is added is 'true'.

* Wait for all the promises to be settled before returning

* fix(SignalingLayer) Update SSRC owners on leave.
Update the SSRC owners in the following cases:
1. When a remote endpoint leaves the call.
2. When a source-remove is received.
3. When a source is remapped (with ssrc-rewriting enabled).
Create the remote track even if presence is not yet received. The ssrc owner check prevents the client from creating a dummy track when the call switches over from p2p to jvb when the last remote endpoint leaves the call.

* ref(SignalingLayer) alpha sort methods.
Clean up unused methods, _findEndpointSourceInfoForMediaType is not used anymore.

* squash: Address review comments.

Endpoint based receiver constraints and other endpoint based bridge signaling messages are no longer supported by latest JVB after the switch to source-name signaling.
Rename method names 'sendNewReceiverVideoConstraintsMessage'->'sendReceiverVideoConstraintsMessage', 'setNewReceiverVideoConstraints'->'setReceiverVideoConstraints'

* ref: Inline onMucMemberLeft (has nothing to do with "moderator" and is broken).

* fix: Use isFocus instead of just reading the resource.

* ref: Inline onParticipantLeft.

* ref: Add _ to private function names.

* ref: Remove redundant function.

* ref: Initialize focusComponent once.

* ref: Remove two copies of "options".

* ref: Simplify the focusJid flow.

* ref: Rename focusCompoonent to targetJid.

* fix: Only retry on "invalid session" errors

We used to fire FOCUS_DISCONNECTED (which doesn't have anything to do
with anything disconnecting) and then retrying. This is not necessary
because FOCUS_DISCONNECTED is a fatal error and the client schedules
a reload of it's own (and we want the client reload as opposed to the
conference-request retry in case e.g. the shard has changed).

* ref: Remove use of visitorFocus.

* ref: Remove graceful shutdown handling (never used in backend).

* feat: Support conference request over HTTP.

* ref: eslint cleanup.

* fix: Add max timeout 2 minutes.

1. Checks peer's preferred codec in p2p case. Mobile and web have different preferred codecs.
2. Log an error message when the preferred codec is not offered by JVB.
3. Clean up code related to deprecated config.js settings 'preferH264' and 'disableH264'.
4. Refactor the codec selection logic so that correct codec is picked.

(Rather than moving streams to inactive in SDP.)

* fix: Do not send empty <nick/> when a nick exists and gets removed

* squash: Fix lint errors.