RTCRTPSender exposes DSCP marking via `networkPriority` in the encodings
configuration dictionaries. That should allow us to control
QoS priorities for different media streams, eg audio with higher network
priority than video. The only browser that implements that right
now is Chromium.
To use this, the public.app.media.networkPriorities configuration in
settings.yml. Audio, camera and screenshare priorities can be controlled
separately. For further info on the possible values, see:
- https://www.w3.org/TR/webrtc-priority/
- https://datatracker.ietf.org/doc/html/rfc8837#section-5
Firefox doesn't fire the ended evt/onended callback for live
MediaStreamTrack(s). We rely on that event.
Manually emit the ended event which works with the onended callback
when a track is stopped
kurento-utils is unmaintained. It's served us well, but its age
shows. We need to transition to something else if we want to
have better maintainability and include simulcast, multistream, ...
This introduces a simplified/leaner wrapper kit that's almost
API-compatible with what we use right now - so widespread changes
are minimal). It's easier to maintain/read/transition from. This
can be read as an intermediate step to transitioning to
something definitive (ie mediasoup-client).
Under some scenarios, cameras are freezing when the virtual background
code is running due to runPostProcessing(_renderMask) throwing
NS_ERROR_FAILURE - mainly on Firefox - consequently preventing
subsequent TimerWorker ticks from being scheduled.
Cases where I've seen that happen are:
- conferences running under an iframe where the iframe is briefly
stalled for some reason
Address the issue with a try-catch and a log for debugability (it's high
frequency, hence why not error level). We should probably remove the log
entirely once we figure out why the post-processing method is failing.
Use the built-in getLocalStream from the peer wrapper instead (which
relies on getSenders - the proper, spec-compliant way).
Two different issues are addressed:
- RTCPeerConnection.getLocalStreams is a pre-1.0 WebRTC spec which is
deprecated and not recommended.
- Fixed an issue where a switch from full audio to listen only could
cause the latter to be rejected with an error 1004 in rare scenarios.
There could be a race condition where the local getDisplayMedia stream ends
(eg via Chrome`s stop sharing toast) while the broker hasn't finished starting.
That would lead to a scenario where the broker wouldn't emit an end event,
causing screen sharing to be flagged as started with a blank/invalid stream.
- Remove the old listen only bridge (kurento.js), superseded by the equivalent
and equally stable (AS FAR AS LISTEN ONLY IS CONCERNED) sfu-audio-bridge
- Rename FullAudioBridge.js -> sfu-audio-bridge.js
* A more generic name that better represents the capabilities and
the nature of the bridge
* The bridge name identifier in configuration is still the same
('fullaudio')
- Remove the FreeSWITCH listen only fallback
- Temporarily disable the "trickle ICE" pair gathering feature used
in SIP.js (which was always experimental, nonstandard and disabled
by default)
- Updates to settings.yml keys in places where relevant
"default" is not an universally valid default value for deviceIds which was causing issues with Firefox and Safari in some specific scenarios where exact deviceId constraints were being used
Seems to have been introduced by a partial merge commit
There were a bunch of style changes introduced by that partial commit as well; I kept those changes to avoid introducing further conflicts between v2.4-2.5...
- forceRelayOnFirefox: whether TURN/relay usage should be forced to work
around Firefox's lack of support for regular nomination when dealing with
ICE-litee peers (e.g.: mediasoup).
* See: https://bugzilla.mozilla.org/show_bug.cgi?id=1034964
- iOS endpoints are ignored from the trigger because _all_ iOS browsers
are either native WebKit or WKWebView based (so they shouldn't be affected)
When joining breakouts, we now wait for the bridge to be loaded before
automatically start user's audio.
This problems happens only on fullaudio bridge
This commit allows user to join/leave audio using the fullaudio bridge.
This is still under development, but to use this now we must set values of
skipCheck to false, and defaultFullAudioBridge to fullaudio. This
depends on newest version of bbb-webrtc-sfu
ICE lite servers (eg mediasoup) dont need candidates signaled out-of-band; neither does KMS in certain scenarios
Disable their signaling saves us some ticks in bbb-webrtc-sfu and some bandwidth all around
Restored the old behavior when ending breakout rooms while user is in the
breakout audio transfer, which is to the trigger the reconnection to the audio
in the main room.
This behavior could be improved by (instead of reconnecting) transfering user
back to the main room, but this requires some changes in akka-apps/fsesl
which can be treated in a different issue.
Closes#13242
Undefined by default means that the governing configuration is in bbb-webrtc-sfu
Also add some inline docs in settings.yml about the media server adapter configs
Applies to video, listen only and screen sharing
New metadata values: media-server-video, media-server-listenonly, media-server-screenshare; parameter is a String
For browsers that don't support headerBytesSent in RTCOutboundRtpStreamStats
neither headerBytesReceived in RTCInboundRtpStreamStats, we are now able
to calculate upload and download rates.
We are also able to get transportStats information for browsers that
don't support iceTransport attribute of RTCDtlsTransport.
Added support for getStats in screenshare's service. This works similar
to the getStats for video provider, and the information retrieved from
screenshare is added to the video information for cameras.
Changes (maybe not a complete list):
- Disable virtualbgs by default
- Move the virtualbg selector in video-preview to the side below the
profile selection
- Restore old video-preview sizes
- Add a wrapper class for MediaStreams (BBBVideoStream)
- Centralize virtualbg services and business logic code into BBBVideoStream
- Refactor and centralize virtualbg constant fetching
- Refactor and centralize virtualbg config fetching
- Organize virtualbg type definitions
- Remove added states in video-provider to prevent further bloat
- Remove added states in video-preview to prevent further bloat
- Lock virtual bg switching while video-preview itself is locked
- Add proper virtualbg error surfacing via toasts
- Refactor iOS availability detection to use centralized UA checker
- Avoid calling gUM when toggling virtualbgs on/off
- Make virtualbg video-list-item action a toggle instead of a
state-aware action
- Make virtualbg switching work in video-preview for cameras that are
already shared. Especially useful when there are multiple source
cameras, and will be important in the near future
- Add Derivative Work notices in files that are partially copied from
jitsi-meet
- Simplify track replacing in video-provider
- Split video-preview UI code for virtualbgs into a separate functional component
