Currently, collection cleanup code is only run when an added event
is received from the server. Where that fails is in scenarios where
a server-side collection turns empty while an affected users is
disconnected - and then reconnects. There's no removed (or updated)
event so no cleanup code is run and you have stale data.
This commit guarantees a stale data check is run whenever a subscription
is established again. The `added` check was also maintained, although
I'm not too sure anymore it's is still needed. That may need to be
revisited.
There are some situations where previously set deviceIds (
local/session storage) may become stale. This causes an unexpected
behavior where audio is temporarily borked until the user clears their
local storage.
This issue has been seen more recently on Safari endpoints when switching
back-and-forth breakout rooms in environments running under iframes.
Also seen randomly on endpoints with virtual input devices.
This centralizes audio gUM calling into a single method that retries the
gUM procedure without pre-set deviceIds only if the initial call fails
due with an OverconstrainedError - hopefully circumventing the issue.
BBBVideoStream 'inactive' event currently does not send the
MediaStream id as the payload for handlers. This can cause theoretical
race conditions due to media stream mismatches in places where
'inactive' handling is done globally (eg video-preview).
This adds the MediaStream id to BBBVideoStream's 'inactive' event
and uses it in video-preview to avoid such race conditions.
Extract the deviceId again from the stream to guarantee consistency
between stream DID vs chosen DID. That's necessary in scenarios where,
eg, there's no default/pre-set deviceId ('') and the browser's
default device has been altered by the user (browser default != system's
default).
There's no rollback procedure in case a device switch fails right now,
nor does the code entrypoints that call the switching procedures wait
for resolution or failure before marking the new device as chosen. That
may cause inconsistent states in a couple of ways:
- No rollback: switch fails, audio is still on but no actual
microphone input is being transmitted
- Not waiting for resolutions: inconsistent chosen devices on failures
Device switching errors are also not surfaced to the end user
This commit:
- Adds device rollback and proper resolution/failure response
awaits to try and make the state a bit more consistent.
- Centralizes the input device switching code to be reused between
different bridges
- Centralizes device ID state management in audio-manager to try and
mantain them a bit more consistent across the board
- Surface device switching failures to the end user
- Guarantee device IDs are set to the session storage on all
appropriate scenarios
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
Here's what we do when user activates mic:
1 - When we do something similar to listenonly's joining process
until we find a valid candidate-pair. The information about this
local candidate is store.
2 - We then start a new userAgent, and as soon as browser finds
a candidate with the same local ip address, we leave only this
candidate in the SDP and send this to FreeSWITCH. SDP should
contain only a single candidate.
3 - The rest of signaling process is basically the same.
Remove parts of a previous connection monitor.
To add some context (as far as my memory goes) to the multiple connection
monitor features the product has, `stats` (currently named `connection status`)
was introduced at the Flash client back in ~2016. @fcecagno and I did it
as a BigBlueButton's Summit activity. Our work was squashed into a single
commit in 92554f8b3e :).
I'm not sure about the whole story behind `network information` (the late
connection monitor added to the HTML5 client) but I assume it should work
as a collector for a bunch of different connectivity monitors. I remember
when it was introduced but I don't know why it wasn't adopted. My best guess
would be because of some performance issues the `user list` had back then.
To follow on why `connection status` replaced `network information` at the
HTML5 client, when I did the `multiple webcams` feature I had to refactor
a big chunk of the `video provider` (#8374). Something that wasn't really
helping there was the adaptation of `stats` that was made to show local
feedback for each webcam connection. Although this feature wasn't being
used anymore, `network information` did rely on that to build up data. With
this monitor gone I assumed it was my responsibility to provide an alternative
so I promoted Mconf's port of the Flash `stats` monitor to BigBlueButton's
HTML5 client (#8579).
Well, that's my perspective on how things went for those features. If
anyone would like to correct me on something or add something else on
that history I would appreciate to know.
This commit contains three fixes: one already reported and two detected
during the investigation of the solution.
This started as a fix for firefox (#12023), but i also fixed the muted
alert/banner when device changes: the banner wasn't detecting device changes,
unless audio was deactived/actived.
There's another fix for the microphone stream: we now keep sender's track
disabled if it was already disabled for the sender's track of the previous
selected device.
Also did small refactor for eslint checking.
Some technical information: in sip bridge (bridge/sip.js), setInputStream and
liveChangeInputDevice function were both fully turned into promises, which
guarantees we have everything ready when it resolves to the respective values.
This helps AudioManager (audio-manager/index.js) to sequentially sets and
tracks the state of the current microphone stream (inputStream), when calling
liveChangeInputDevice function: we first set the current stream to null,
creats a new one and then set it to the newly created value - this is needed
because MutedAlert (muted-alert/component.jsx) can then gracefully
allocate/deallocate the cloned stream when it is set to a non-null/null value
(the cloned stream is used for speech detection with hark).
In MutedAlert we also make sure to enable the cloned stream's audio
tracks, just in case the user change the device when muted (audio track is
disabled in this case), which also leaves the cloned stream muted (we then
enable the track to allow speech detection).
Closes#12023
Firefox doesn't create a device called 'default' and we were trying
to set this when user is joining the room. We don't do this anymore, letting
devices to be changed when there's some user request.
Moved outputDeviceId inputDeviceId information to be managed in bridge
(just like we do with inputDeviceId), we don't store this duplicated
information in audio container anymore.
Fixed the eslint warning in "playAlertSound(url) { ..."
We are safe to let users try to change input/output devices because the
device list is retrieved from enumerateDevices.
