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'.

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.

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

When a msid attribute is missing in the 'a=ssrc' line, use the local endpoint id as an indentifier. Move this generation logic to LocalSdpMunger. Also suppress notifying Jicfo of a ssrc change on Firefox when the change is a result of the media being suspended on the jvb connection.

Get rid of noisy logs related to SDP transformations which are redundant. Fix formatting and add missing information.

In certain cases, when RTCPeerconnection#addTrack is used, the browser doesn't produce the streamid in the 'a=msid' line which is being used by Jicofo as a unique identifier for the associated ssrc. Whenever the msid is missing in the local description, generate an ID using the local endpoint id.

SDP utility classes are spread across RTC and XMPP directories now, moving these class files to a 'sdp' directory.

Disable simulcast for low fps screenshare and enable it for high fps screenshare. testing.capScreenshareBitrate config.js setting has now been deprecated.

This will reduce the numbers of transport-info IQs sent by the client.

Make sure the codec order is munged on all the m-lines for unified plan clients. Implement the logic for setting the preferences through RTCRtpTransceiver#setCodecPreferences.

* fix(TPC): Configure degradation preference in RTCRtpSendParameters.
Properly configure degradation preference on RTCRtpSendParameters instead of RTCRtpEncodingParameters. Fixes https://github.com/jitsi/lib-jitsi-meet/issues/1510.

* feat(JingleSessionPC): Remove ssrcs from remote desc when a user leaves.
Remove the ssrcs (associated with remote sources) from the remote desc along with the removal of the remote tracks when an endpoint leaves the call. The source-remove signaling message from Jicofo will no longer be needed in this case and can be dropped.

All the send video constraints for the client, i.e., what simulcast streams will be enabled based on constraints received on the bridge channel, will be handled by the SendVideoController class.
The receive video constraints like lastN, selectEndpoints and receive video resolution will be handled by the ReceiveVideoController class.

Force the client to reload when the bridge that is handling the media goes down.
This mitigates issues seen on the bridge because of a client re-joining the call with the same endpointId, BWE issues, etc.
This behavior is configurable through 'enableForcedReload' setting in config.js.

RN doesn't support RTCRtpSender yet. Therefore, media is suspended on RN by changing the media direction in the SDP whenever the client receives an ideal height of 0 for sender constraints on the bridge channel.

After ICE failure and subsequent recovery of the media connection to the bridge using session-terminate, the local connection status needs to updated when ICE connection is established on the new peerconnection. The local connectivity status will be shown as Lost otherwise.

Determine the preferred codec for a given endpoint based on the config.js settings and the codecs supported by the endpoint.
The preferred codec is published in presence and then used by the other endpoints in the call during join/leave to determine
if the codec needs to be changed on the fly. Different codecs can be configuered for p2p/jvb connections.
The preferredCodec/disabledCodec settings under videoQuality will have precedence over the older settins like preferH264/disableH264.

Configure the encodings on FF after the track is added.

Calculate the bitrate based on the sender video constraint applied on the track and the bitrates specified for different resolutions.

Use 'preferredCodec' and 'disabledCodec' under videoQuality config.js settings for setting codec preferences.
Do not prefer VP9 on Firefox because of https://bugzilla.mozilla.org/show_bug.cgi?id=1633876.

Make sdp munging logic for changing codec preferences more generic
Do not negotiate High profile H264 codecs on when RN clients are in the call

Handle the promise returned by RTCRtpSender.setParameters.

* feat: Apply max bitrates on video sender
Add the ability to control the max bitrates on the video sender through a config.js videoQuality setting.

* ref: Use '_' prefix only for internal methods

Delays e2e encryption initialization until any encryption key is set.
Recreates peerconnections in order to enable insertable streams
only if the e2ee is used. This is to avoid bug around insertable streams
which may cause audio issue when the main JavaScript thread is loaded:

https://bugs.chromium.org/p/chromium/issues/detail?id=1103280

* feat: Set degradation preference on video senders
Use 'maintain-framerate' for camera tracks and 'maintain-resolution' for desktop tracks

* ref: apply cached video contraints and degradation preference on unmute
Since the sender encodings are available only after the renegotiation is complete (on chrome/safari), constraints/preferences have to be applied after renegotiation is complete.

Refactors the way send/receive video constraints are managed and puts the high level
logic in a separate module. See QualityController.js for high level overview on how
the constraints are managed now.

Adds events to JitsiConference fired whenever it starts new jvb/p2p session.

Adds event to JingleSessionPC.js when remote party signals receive max frame height.
Also adds signaling of the local recv preference for the p2p mode(only existed for JVB).

Switches from using JitsiConference.isJvbConnectionInterrupted
flag to peerconnection's ICE state. The interrupted
flag is not set if the connection fails initially.

Audio fails on Safari when we try to re-use audio m-lines whose direction has been set to 'inactive'
when the remote source left the conference.
Disable noisy mic detection on Safari as this causes audio input to fail on Safari on iPadOS