* Refactor: Remove the old user status (reaction/emoji)
* Update settings.yml
* test: remove old user status testing code related
---------
Co-authored-by: Anton B <antonbsa.bck@gmail.com>
* feat(html5): initial implementation of Gladia transcriptions to BBB 3.0
* fix(transcription): Add missing locales and fix invalid cc menu key
* fix(bbb-transcription-controller): Bump transcription controller to fix some bugs
* fix: adjust yq syntax for setting fs esl password in transctiption-controller
* fix(transcription): Use newer useSettings format from transcription options
* fix(captions): Correctly use captions settings
---------
Co-authored-by: João Victor <joaovictornunes973@gmail.com>
Co-authored-by: Anton Georgiev <anto.georgiev@gmail.com>
Co-authored-by: Ramón Souza <contato@ramonsouza.com>
* refactor(storage): replace Tracker.Dependency with observer hook
* fix(storage): set initial value
* refactor(storage): stop using Meteor's Session singleton
The audio troubleshooting modal has very microphone-specific strings,
which might confuse users trying to join listen only.
Review the Help screen so that listen only scenarios are more generic.
As a bonus, review the unknownError locale with a more actionable text.
Listen only has a built-in retry routine on join failures that's
convoluted half-broken. It stems from the Kurento era where it could
fail randomly due to a myriad of reasons.
Production logs indicate that the retry is seldom used nowadays in
mediasoup-based environments. The presence of the retry also breaks
the error troubleshooting modal when actual failures happening, leaving
users in the dark about what's happening.
Remove the listen only retry code from AudioManager and bubble up any
join failure to the callers.
In scenarios where the join audio flow skips echo test, NotAllowedError
(and any other errors) are all being mashed together under a generic
MEDIA_ERROR object.
Properly handle specific errors in audio-manager so they're correctly
render in the audio modal help screen.
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.
