* fix(JingleSession): Avoid renegotiation when user with no sources leaves the call.
* feat: participant kick reason add
* ref(RTC): remove legacy pc constraints. Stop using the legacy pc constraints that are no longer wired up to WebRTC.
* fix(deps) update webrtc-adapter to v7.7.1
087a8e19eb...4191198233
* squash: Use different function syntax.
* squash: Fix lint errors.
* Process stats immediately before setting the interval.
* feat(ReceiveVideoController): Add the ability to send constraints in the new format. Add the ability to send the bridge messages for the receiver video constraints in the new format directly.
676c7a9105...5796d83bb1
* feat(browser-support): Add support for WKWebview based browsers. 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.
08ce96d881...e60f09b189
* squash: Always get lastN value from JitsiConference instance.
* fix(lastN): Return the correct lastN value for the conference.
* Use unified plan for mobile browsers on iOS
d31b5a2d5e...08ce96d881
* fix(conference): Do not signal muted tracks on join. 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).
e83fb93d2d...d31b5a2d5e
* fix(RTC) fix device selection not being available
* fix(TPCUtils): undefined is not an object (evaluating 'this.tpcUtils.replaceTrack(e,t).then')
4c668023b3...e6ef4e7ae9
* 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...4c668023b3
* 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...6a7b16c33e
* 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...7f919faacc
* ref(QualityController): Split send and receive video constraints handling.
* fix: Save guards _features to be always empty and nver undefined. (#1493)
d1f0ab4d5a...c534f74884
* 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...d1f0ab4d5a
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.
479dd98...77978f0.
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...be18ff34be.
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.
Jaya Allamsetty
dependabot[bot]
Saúl Ibarra Corretgé
hmuresan
Hristo Terezov
Andrei Gavrilescu
Alex Bumbu
Дамян Минков
Steffen Kolmer
Tudor-Ovidiu Avram
Tudor D. Pop
Jaya Allamsetty
Vlad Piersec