Get rid of the browser specific keys and use the standard spec-compliant fields for stats.
Get the resolution/fps for remote streams from 'inbound-rtp' stats. Use the 'track' stats for the local resolution/fps since these take the active simulcast streams into account.

Remove the effect instead of stopping it so that the original stream is restored on both the local track and on the peerconnection. Fixes issues when a stream with effect applied is replaced on the pc after it is muted, also fixes https://github.com/jitsi/lib-jitsi-meet/issues/1537.

Calculate the target bps based on the video quality settings and the codec configured on the peerconnection.
Hardcode target video bitrates for RN since it doesn't support setting max bitrates.

* feat: Video type camera is default value of missing, skip it in presence.

We skip sending initial video type camera if it is already missing from presence, if it changes we send the last value.
We also are dropping the namespace as it is not used anywhere and just increases presence size.

* feat: A/V muted is default so we can skip adding it initially to the presence.

Dropping the namespace as it is not used anywhere and that reduces size.

* squash: Drops unused setting.

* squash: Adds a config option to enable skipping the muted state.

* fix: Adds back removed method used by jibri.

It was removed by https://github.com/jitsi/lib-jitsi-meet/pull/1512.

* squash: Adds a deprecated note.

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.

Use the new Colibri message "EndpointStats" for broadcasting the local stats. The bridge then will be able to filter the endpoint stats and send them only to the interested parties instead of broadcasting it to all the endpoints in the call.

 RTCRtpReceiver.getCapabilities could be undefined on some platforms

Drop end to end signature. The impersonation attack vector is deemed out of
scope since insiders are already part of the meeting.

Capture the audio levels only for the top 5 speakers as RTCRtpReceiver#getSynchronizationSources can be expensive when we have too many audio receivers in the call.
Also, capture the audio levels for track that are unmuted if RTCRtpReceiver#getSynchronizationSources is not supported.
Switch Safari to using getStats since its reporting errorneous values, i.e., 0.000001 as audio level for all remote audio tracks.

* Feature: Moderator can revoke moderator role to others and himself

* Code review inputs

* Code review inputs

Stop using the legacy pc constraints that are no longer wired up to WebRTC.

Fix Safari RTCSessionDescription.sdp readonly
Based on https://bugs.chromium.org/p/chromium/issues/detail?id=1186456&

* 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.

Add the ability to send the bridge messages for the receiver video constraints in the new format directly.

Apple added getUserMedia support for WkWebview based browsers like chrome and Firefox on iOS 14.3. These browsers behave as Safari does on iOS. Therefore, extend the Safari checks to these webkit based browsers as well.

All browsers on iOS use WebKit engine so they use the Unified Plan.

Do not add the muted audio/video tracks to the peerconnection on join. The tracks will be added when the user unmutes for the first time. This reduces the number of remote sources that will be added when a participant joins a large call where everyone joins muted (startAudioMuted/startVideoMuted setting).