There's a plan for using ping to detect XMPP WebSocket
disconnected instead of waiting for timeouts on the network layer.
This also simplifies ICE failed handling logic.
...in setMediaTransferActive. The sender video constraints
are controlling the 'encoding.active' flag and this code
would interfere. It's right now not used anyway as P2P is
disabled in the unified mode. If this modification is
necessary we'll add it back when working on unified P2P
in such a way that will play nicely with the sender constraints code.
...when starting without video.
The app would crash in setSenderVideoConstraint,
because the RTP encoding are not available on Safari
until sLD/sRD cycle is performed.
ref: Moves xmpp logs to be accessed from connection. (#1284)
* ref: Moves xmpp logs to be accessed from connection.
In cases where there is no room like pre-join and lobby screen we still want to be able to debug xmpp messages.
* squash: Remove getLogs from Conference.
* squash: Fix catch.
* squash: Print the error.
fix(video-quality): Get downsampled video for p2p case
When the video sender constraint changes in the p2p case when the remote user switches between stage/tile view,
get a downsampled video of the original 720p stream instead of requesting the camera for a new stream with the lower resolution.
Fixes https://github.com/jitsi/jitsi-meet/issues/7267
* 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
Add a performance stat around long tasks. Chrome supports PerformanceObserver API that lets us
register for long tasks event. Any task that takes longer than 50ms is considered a long task.
fix: Scale remote audio levels reported on receiver to getStats levels
The audio levels reported on the audio receivers are lower when compared to the value reported by getStats.
Values reported by getStats on chrome do not follow the the spec and since we have combination of clients using both getStats and getSynchronizationSources,
lets stick to one scale to make them look uniform.
Also, the receivers seem to be reporting audio level for a little bit after the remote user has muted. Make sure the track is unmuted
before setting the audio level on the track.
fix(StropheLastSuccess): refresh the timestamp in connected
An 'item-not-found' or other error on Websocket reconnect will refresh
the timestamp which will give misleading information about when
the last stanza was received.
fix(JitsiConference): skip participant kicked event for kicker
With the introduction of lobby the ChatRoom KICKED event is now also emitted for
ourselves (the kicker) so we want to avoid emitting an event where `undefined`
kicked someone.
Remove redundant _channelOpen flag from RTC and rely on the channel's isOpen() getter instead.
This fixes an error throw when the _channelOpen flag was not updated while the channel's been
closed by the bridge.
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
fix: RTC track events emitted when switching jvb<->p2p
JitsiConference should emit JitsiConference events instead of RTC events
when adding/removing tracks on jvb<->p2p switch. This behaviour was mixing RTC layer
with the conference one. When conference switches from jvb to p2p it wants to make
the application switch to use p2p JitsiTracks, but it doesn't mean that the JVB tracks
have been removed on the RTC/peer connection level.
feat: use getSynchronizationSources on the receiver for remote audio levels (#1245)
* feat: use getSynchronizationSources on the receiver for remote audio levels
Use getSynchronizationSources if it is supported, fallback to using getStats otherwise.
* feat/ref: Use the local audio levels from LocalStatsCollector
When using getSynchronizationSources, use the audio levels from LocalStatsCollector for NoAudioSignalDetection.js
Remove obsolete code - TalkMutedDetection feature using audio levels is not used anymore
Takes disabled encodings into account when calculating the local resolution. (#1242)
* fix: Takes disabled encodings into account when calculating local resolution.
* feat: Adds a new event that's triggered when the max enabled resolution changes.
* feat: Broadcasts the max enabled resolution value along with other stats.
paweldomas
Jaya Allamsetty
Christoph Wiechert
damencho
Saúl Ibarra Corretgé
Jaya Allamsetty
Vlad Piersec
dependabot[bot]
Boris Grozev
George Politis