It creates its own emitter now extending Listenable.

This is breaking xep-0045, but this is just between the client and prosody and will be added only when we send the initial conferenceIq via http.

More than one VP9 payload type is produced in the offer, more experimental ones if the local endpoint has no local video source. Munge the SDP to remove all codecs besides the one with profile-id=0 so that only one VP9 codec is negotiated with p2p peer.

This is much faster than changing direction on the transceiver and doesn't need a renegotiation for the suspending/resuming media. Also fixes an issue with audio-only mode in p2p where media doesn't resume when audio-only mode is disabled if it was enabled while the jvb connection was active.

For renegotiations triggered by the application, it is ok to execute sRD->cA-sLD cycle on the p2p initiator instead of executing cO->sLD->sRD cycle. Chrome sometimes creates a third m-line for p2p when createOffer is called which is unexpected and therefore it breaks the client's renegotiation cycle. This has been done in other places already, i.e., when remote sources are signaled, etc. so making it uniform across all renegotiations.

Fixes an issue when a screenshare doesn't show up on remote endpoints after one of the endpoints in the call does an ICE RESTART using session-terminate while it has 2 video sources.

* feat: Handles join error about required display name.

* squash: Fixes tests.

* squash: Adds param to DISPLAY_NAME_REQUIRED event indicating is it coming because of lobby.

* squash: Drops all clear up methods for unrecoverable errors.

The jitsi-meet part is taking care of it.

supportsCodecPreferences is a function, and must be called

* 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.

Fixes blurry screenshare in some cases. There is a higher probability of the higher layers getting suspended when all the stream encodings are enabled for low fps SS. Make sure only the highest spatial layer is sent for low fps SS after p2p->jvb switch. Make sure all the stream encodings are enabled for high fps SS after p2p->jvb switch.

Check for source updates and notify the peer of any new sources after every renegotiation cycle. This fixes issues where source changes are not detected if any other task that requires a renegotiation is scheduled between replaceTrack and a negotiationneeded event.

Fixes a bug where the remote video freezes if the remote user does an ICE restart using session-terminate. In this case, a new SSRC is assigned to the remote source which will have the same source-name.

Pull initial offer/answer tasks out of the modification queue and execute them right away. Only track and codec related operations that necessitate a renegotiation cycle need to be pushed to the modification queue. The queue execution is paused until the session is established. This avoids track operations being executed before the session is established. This fixes an issue seen in p2p connections where sources of the initiator are not signaled to the remote since the tracks are added while the initiator is waiting for a session-accept from the peer.

With visitors we rarely see not able to send messages or presence because not connected error.
Seems the status of the connection is wrong when quickly disconnecting and connecting again.

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

Any jibri recording. streaming or sip jibri will join the main room.

When receiving error like max occupants reached we have not joined the room, and there is no point of sending presence unavailable and check and clean after 5 secs.

When configureSenderVideoEncodings task is pushed to the modification queue, there might not be a setParameters call in progress and this new one will not get chanined to any promise. However, when the task finally is executed, there is a chance for another setParameters method execution to be in progress and can result in setParameters call failing. Also, these extra steps to ensure that the sender encodings are correctly encoded are no longer needed since with multi-stream mode, we do not change the source video type and the previous encodings config will still be valid.

* Add a feature flag and xmpp-caps entry for joining as a visitor.

* Default to enabled, use for the confernece-request flag too.

* feat(visitors): Visitors chat.

For messages coming from visitors, which are not in the current room we need to supply a display name.
Mark messages with custom display name as coming from guest.

* feat: Adds a visitors version to conferenceIQ.

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.