Update notes:

- The base64-sha1 code got stripped out, so we bundle it now
- There are 2 optional dependencies which we want to ignore to avoid
  extra bundle size: xmldom and ws (the former is used by RN but we
  provide it in the app and the latter is only used by Node
  environments)

* feat: Add initial-last-n to jingle if config.channelLastN is set.

* Also add initial-last-n to transport-accept.

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

* feat(ssrc-rewriting) Advertise support to Jicofo.

* Fix unit test

* 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.

When the application doesn't set receiver constraints, generate default constraints in the source-name mode.
Cleanup code related to the endpoint based video constraint handling for p2p connection.

Simulcast for SS is always on in unified plan mode, the lower spatial resolution streams are turned off for low fps sharing.

Always initiate a sRD->cA->sLD cycle since renegotiation fails in the following scenario.
In a p2p call when channelLastN=0, the direction on the video tranceiver is set to'inactive'.
At this point, if the user unmutes, the track is replaced on the video sender.
If a cO->sLD->sRD is triggered, the browser adds a third m-line which isn't expected and possibly is a bug. All renegotiations fail as a result.
However, the browser does not add a third m-line in the answer it generates and renegotiation succeeds.

* 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.

<conference xmlns="http://jitsi.org/protocol/focus" vnode="v1" ready="true" room="roomtest@conference.focus.exmaple.com" focusjid="focus@auth.meet.jitsi">
    <property name="authentication" value="false"/>
    <property name="sipGatewayEnabled" value="true"/>
</conference>

* Remove callstats conference property.

* Remove RTX conference property (removed from backend).

* Remove audioPacketDelay conference property (removed from backend).

<iq id='some-id' from='focus.exmaple.com' type='error' xml:lang='en-US' to='some-jid' xmlns='jabber:client'>
    <error type='modify'>
        <redirect xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/>
        <conference xmlns='http://jitsi.org/protocol/focus' vnode='v1' main-room='roomtest@conference.focus.exmaple.com' focusjid='focus@auth.meet.jitsi'/>
    </error>
</iq>

* feat(face-landmarks): add timeline

* add duration to xmpp payload

* Remove support for legacy endpoint based signaling and make source-name signaling as the only signaling mode. The legacy screensharing mode will no longer be supported. The client will still be able to process remote presence sent in the old format to support interop with very old mobile clients and jigasi that do not support source-name signaling.
* remove code related to presenter mode.

The candidates are not necessary when we use jvb.

Instead of Jicofo signaling all the remote sources available in the call, the bridge now signals only a limited set of SSRCs and then rewrites the SSRC on the outgoing media streams. The SSRC mapping is done based on the sources requested by the clients through the receiver constraints. This limits the number of m-lines in the remote/local SDPs on the client and therefore results in better performance in large calls.

* Handle source remapping messages from bridge
* Added track_owner_changed events
* don't process an invalid rtx ssrc.
* keep track of remote ssrcs, only renegotiate on new ones.
* Change source name on remote track on ssrc remapping.
* Don't remove tracks on member leave.
* Remove (orphaned) tracks on session terminated.
* Use serial number (per media type) to create msid attribute.
* Update videoType on remapping.

Co-authored-by: James A <jqdrqgnq@users.noreply.github.com>

* fix(codec-selection) Munge all related m-lines for multi-stream.
Munge all the video m-lines to apply the preferred codec settings in multi-stream mode.

* fix(TPC) Cache transceiver mids for all local video tracks.
There will be multiple video tracks in multi-stream, we need to cache the mids for all the the associated transceivers.

* fix(sender-constraints) Do not apply sender constraints if value hasn't changed for a given source.

* fix(TPC) Configure the direction correctly for all m-lines.
Ignore negotiationneeded events for p2p to avoid going into a renegotiation loop.

* squash: Address review comments.

Fixes cases where SS fails in multi-stream mode with p2p->jvb->p2p switching.

Avoid needing an external dependency.

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>

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

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.