WebRTC-based stats generation in the connection status modal is broken
on Firefox >= 125. A broken type check coupled with a new partially
implemented RTCIceTransport dictionary causes and undefined function
call when fetching the selected candidate pair. Since that error is
unhandled, collection breaks.
Correctly check for the getSelectedCandidatePair method availability in
RTCIceTransport so that it skips to pair inference from getStats if
necessary.
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.
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.
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.
