There's a race condition that may cause a client crash whenever a
connectionstatechange callback is cleaned up in a peer without a
valid peer connection present in our custom RTCPeerConnection wrapper.
Check for peerConnection availability in the WebRtcPeer wrapper before
trying to clean up its connectionstatechange callback.
A client crash may happen if either the Meeting collection or the
document's metadataProp attribute are undefined whenever the
getFromMeetingSettings util is called to fetch metadata.
It's debatable whether anything is working in the client if the
documents being accessed here are unavailable, but it'll still be logged
and might bork an ongoing reconnect.
Use optional chaining + nullish coalescing to avoid causing TypeErrors
in those situations while also returning default metadata values
properly.
When a sendrecv peer acts as the answerer, gUM is only called _after_
the remote offer is received. This is fine, but the error handling runs
different in that scenario in a way that eventual gUM errors are treated
as negotiation errors, leading to inconsistencies when surfacing the
error to end users.
If a peer is acting as answerer and is a transceiver, acquire the local
streams _before_ actual negotiation so that gUM errors are surfaced
correctly (and we spare uneeded negotiation steps).
Audio exit toasts are fired in some redundant situations, e.g.: when the
error help screen is toast.
Change the logic a bit so that it's only fired when the audio help modal
won't be shown, i.e.: when audio had succesfully connected.
* Demo changes
* Revert "feat(captions): no longer writes in the pad"
This reverts commit a76de8c458.
* feat(transcriptoin): Add config options for the transcription backend
* feat(transcription): Add autodetect option to cc chevron
* feat(transcription): Move transcription options into settings modal
* feat(transcription): Set transcription options via userdata
* fix(transcription): Correct userdata for settings transcription params
* feat(transcriptions): options to auto enable caption button
* feat(transcriptions): Option to hide old CC pad funcionality
* fix(transcription): Fix PR comments
* fix(transcription): Refactor updateTranscript to prevent null user and make it more readable
* feat(transcription): bbb_transcription_provider can be set via userdata
* fix(transcription): Use base10 for parseInt
* fix(transcriptions): Fix CC language divider when using webspeech
* fix(transcriptions): Use a default pad in the settings instead of hardcoding 'en'
We still need to use a language pad such as 'en', but in the future we can better
separate these systems.
* fix(transcription): Add a special permission for automatic transcription updates to the pad and restore old per user updates permission
* feature(transcriptions): Include transcriptions submenu and locales
* chore: bump bbb-transcription-controller to v0.2.0
* fix(transcription): Add missing menu files
* fix(transcription): Fix transcription provider options in settings.yml
* fix: setting password for bbb-transcription-controller
* build: add gladia-proxy.log for transcription-controller
* fix(transcriptions): Remove transcript splitting and floor logic from akka apps
* fix(captions): Show long utterances as split captions, show multiple speaker captions
* chore: bump bbb-transcription-controller to 0.2.1
---------
Co-authored-by: Anton Georgiev <anto.georgiev@gmail.com>
This is an initial, experimental implementation of the feature proposed in
https://github.com/bigbluebutton/bigbluebutton/issues/14021.
The intention is to phase out the explicit listen only mode with two
overarching goals:
- Reduce UX friction and increase familiarity: the existence of a separate
listen only mode is a source of confusion for the majority of users
Reduce average server-side CPU usage while also making it possible for
having full audio-only meetings.
The proof-of-concept works based on the assumption that a "many
concurrent active talkers" scenario is both rare and not useful. With
that in mind, this including two server-side triggers:
- On microphone inactivity (currently mute action that is sustained for
4 seconds, configurable): FreeSWITCH channels are held (which translates
to much lower CPU usage, virtually 0%). Receiving channels are switched,
server side, to a listening mode (SFU, mediasoup).
* This required an extension to mediasoup two allow re-assigning producers
to already established consumers. No re-negotiation is done.
- On microphone activity (currently unmute action, immediate):
FreeSWITCH channels are unheld, listening mode is deactivated and the
mute state is updated accordingly (in this order).
This is *off by default*. It needs to be enabled in two places:
- `/etc/bigbluebutton/bbb-webrtc-sfu/production.yml` ->
`transparentListenOnly: true`
- End users:
* Server wide: `/etc/bigbluebutton/bbb-html5.yml` ->
`public.media.transparentListenOnly: true`
* Per user: `userdata-bbb_transparent_listen_only=true`
Add secondsToActivateAudio, inputDeviceId, outputDeviceId and isListenOnly
to audio_joined.extraInfo
Add inputDeviceId, outputDeviceId and isListenOnly to
audio_failure.extraInfo
Add a try-catch to the device enforcement procedure triggered by
onAudioJoin - it may throw and block the modal.
In peer.js offer generation method, the provided constraitns are only
being type checked. It needs a value check when the constraint is a
boolean.
This commit should prevent useless transceivers from being added when
explictly specifying audio/video as `false`
There's an edge case in finnicky networks where ALG-like firewalls
tamper with USE-CANDIDATE STUN packets and, consequently, bork ICE-lite
connectivity establishment. The odd part is that client-side gathering
seems to complete if intermediate STUN bindings work (before the final
USE-CANDIDATE), which may cause the peer not to generate relay
candidates == connectivity fails.
This adds the `public.kurento.gatheringTimeout` option to forcefully extend
the candidate gathering window in peers that act as offerers. The
behavior is as follows: if the flag is set (ms), the peer will wait
either the gathering completed stage or, _at most_,
public.kurento.gatheringTimeout ms before proceeding with calls chained
to setLocalDescription.
This option is disabled by default and intentionally ommited from the
base settings.yml file as to not encourage its use. Don't use it unless
you know what you're doing :).
