* fix(RTC) fix device selection not being available
* fix(TPCUtils): undefined is not an object (evaluating 'this.tpcUtils.replaceTrack(e,t).then')

4c668023b3/...e6ef4e7ae974e054d6647d05718c18f678ed3e9c

* fix(TPC): Remove the existing track instead of overwriting. When a second remote track of the same mediatype is received for an endpoint, remove the existing track before creating the new remote track.

9beb47fe5f/...4c668023b38c92615cd06c4e90006246c3e4aa2c

Fixes: https://github.com/jitsi/jitsi-meet/issues/8683

We use TFLite now so this is unused.

* Added mute video feature

* Fixed export

* Fixed some issues

* Added remote video mute notification

* Fixed import

* Fixed conference event handling

* Fixed some linting issues

* Fixed more linter errors

* turn screenshare off on remote video mute

* Fix linter issue

* translations added for mute video feature

* Added video mute button to interface config

* Updated lib-jitsi-meet

* Fix copy paste error

Co-authored-by: nurjinn jafar <nurjin.jafar@nordeck.net>

* fix(e2ee) fix disabling E2EE
* fix(e2ee) fix key index after ratchetting
* fix: Drop caps handling (#1495)
* fix(SendVideoController): Apply the sender constraint only when it changes. There were cases where the bridge was sending the same constraint multiple times causing redundant calls to getParameters/setParameters on the RTCRtpSender.
* feat: Use the new bridge signaling format.
* fix(gum) update permissions prompt detection

c534f74884/...6a7b16c33e27481b03e5a37636e72426c0848265

Show an overlay with a spinner when slow gUM is fired

* fix(SendVideoController): Apply the sender constraint only when it changes. There were cases where the bridge was sending the same constraint multiple times causing redundant calls to getParameters/setParameters on the RTCRtpSender.
* fix(gum) update permissions prompt detection

beaff3dd02/...7f919faaccb268ef307d619992260919a6535e95

* feat: Use the new bridge signaling format.

c534f74884/...beaff3dd02e9e99da2f61c2a9ed5ca3a34d51dcf

Use the Google Meet model and SIMD optimized WASM.

* ref(QualityController): Split send and receive video constraints handling.
* fix: Save guards _features to be always empty and nver undefined. (#1493)

d1f0ab4d5a/...c534f748849a308d08b06e306f5a66709ccae056

* fix(GUM-permissions): cache permissions on init.
* feat: Reuse billingId from localstorage as jitsiMeetId.
* fix(example) simplify
* feat(docs) mvoe API documentatrion to the handbook

84357ce1a8/...d1f0ab4d5a691f438f7d4c7446f29273b1b8fcdc

* fix(GUM): improve permissions logic.
* feat(GUM): timeout.
* ref: Remove pinEndpoint. (#1440)

30c8795770/...84357ce1a8d06c67a502709877130520aaf71592

Add the ability to configure different max bitrates for VP8 and VP9.
Set max bitrate for presenter to 2500 Kbps irrespective of the configured max bitrates for video.
https://github.com/jitsi/lib-jitsi-meet/compare/479dd98...77978f0.

* send meeting uuid to rtcstats

* change ret description

* fix flow error

* update lib-jitsi-meet version

* feat: Adds option to set ws keepalive url through config.

* chore(deps) lib-jitsi-meet@latest

* feat: Adds option to set ws keepalive url through config.
* fix VADAudioAnalyser: catch error

be18ff34be/...9fdde46694d1c4bc8b7f051cc8d50e0df29ffd07

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.
LJM update - 3570339360/...be18ff34bedf38c7475fe4953074c7959000e15f.

831716c160/...3570339360763fa0911d92c5ae7c270860b6934b.

Support processing stats in the spec-compliant format.

* feat: Uses mod_external_services supporting urn:xmpp:extdisco:2.

The old mod_turncredentials.lua is left to continue working for those using old installs.
New install will start using the new module which will no longer be needed with prosody 0.12.

https://hg.prosody.im/prosody-modules/file/4841cf3fded5/mod_external_services/mod_external_services.lua

* squash: Updates ljm to support urn:xmpp:extdisco:2.

https://github.com/jitsi/lib-jitsi-meet/compare/...94ac35ae818093896e639e74f5fc389b488206a0

* feat: Detects shard changed when using websockets. (#1462)

1009693f2e/...cb484cf48ccd3d56cf9bb2e9d452bd7aa644ac8a

    When an endpoint that doesn't support the preferred codec (VP9) joins a conference, all the other endpoints fallback to VP8 until the endpoint leaves the call.

Stats timestamps were incorrectly formatted, fixed upstream.

Fixes: https://github.com/jitsi/jitsi-meet/issues/8367

* fix: Disabling lobby when using tenant.

87c6e37475/...7896fc8b92f1416f800b56c87e51fb69700d620d

Will now work even if installreferrer or GCM are missing.

* feat: Skip p2p when the participant is jigasi.

1fd7256553/...87c6e374755718fdb0804c2c798ea4bc832f4fca

Fixes icons not rendering on iOS 11.

*  feat: Skips using disco-info for features. (#1450)
* sdp: improve sdp matching for simulcast lines (#1452)
* sdp: add missing colon to findLines calls (#1447)

310983c5b0/...1fd725655357db161d40e7dd52ef111e2b04bd68

The amplitude-js library gained React Native support so there is no need to keep
separate implementations.

* fix: Skip sending multiple times disco-info to jicofo.
* build(deps): bump ini from 1.3.5 to 1.3.7

9f65e8fab3/...8bb653f1d6d450f4d17f56eb6bd4c353138f6829

Bumps [react-native-webview](https://github.com/react-native-community/react-native-webview) from 10.9.0 to 11.0.0.
- [Release notes](https://github.com/react-native-community/react-native-webview/releases)
- [Changelog](https://github.com/react-native-webview/react-native-webview/blob/master/.releaserc)
- [Commits](https://github.com/react-native-community/react-native-webview/compare/v10.9.0...v11.0.0)

Signed-off-by: dependabot[bot] <support@github.com>

chore(deps) lib-jitsi-meet@latest

Helps with testing turn relay cases.

chore(deps) lib-jitsi-meet@latest

Also, default to libravatar.

Closes: https://github.com/jitsi/jitsi-meet/issues/4927

chore(deps) lib-jitsi-meet@latest

Safari 14.1 has a bug where it returns 720p for every simulcast stream when RTCRtpSender.getParameters is called even though the stream resolutions are different.
By using the encodings config used when source was added, on every RTCRtpSender.setParameters call, we ensure that simulcast stream resolutions don't change.
chore(deps) lib-jitsi-meet@latest

Do not resize the desktop share to 720p by default when the desktop track resolution is higher than 720p. This is causing bluriness when presenter is turned on.
Remove the 'detail' contentHint setting for the desktop+presenter canvas stream as it forcing chrome to send only 5 fps stream for high resolution desktop tracks.
Move the desktop resizing logic behind a config.js option - videoQuality.resizeDesktopForPresenter.