* 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.

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'

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

It was behind a testing flag setScreenSharingResolutionConstraints.

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.

That's a graceful close by the bridge.

Reorganized the logic a bit to make it more clear.

* 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.

It now supports the sender encodings API.

* fix(BridgeChannel): prevent multiple websockets on retries

* fix(BridgeChannel): ignore different websocket close event

* fix(BridgeChannel): Keep onclose log

* ref: Add comment for retry timeout on connecting ws

* ref: Use logger debug instead info

(Rather than moving streams to inactive in SDP.)

<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).

Revert this change as this is causing a dummy track to be created when the media resumes on jvb connection after the p2p peer has left the call.

Currently, remote tracks are not created if presence for the endpoint is not received before source signaling. With ssrc-rewriting, the source information will be received on the bridge channel and presence on the prosody ws so there are chances that they can be out of sync. We do not want to skip remote track creation when that happens.

<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>

It implements SAS verification as per the Matrix spec, adapted to our environment.

* feat(face-landmarks): add timeline

* add duration to xmpp payload

It's emitted when remotely closed / failed.

* 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.

Use the participant ID as the key to codecs, etc, instead of the source
name. The object is further indexed by SSRC.

Don't wait for both codecs to be set before propagating codec
information.

Sample data model for codecs:

codec: {
    participant1: {
        ssrc1: {
            audio: 'opus',
            video: undefined
        },
        ssrc2: {
            audio: undefined,
            video: 'VP8'
        },
        ...
    }
    ,...
}

Some values, like codecs, are available right away, we don't need to
compute differences.