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
- Return to the ResizeAndMoveSlide event to do pan&zoom, respecting the viewed width and height ratio
- Defaults zoom in toolbar to 100% like before to be more consistent
- Fit to width and Reset Zoom is back (fit tho width still has some sync problems)
- Fix to not change to first page when presenter reloads page
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
Move the language collection to the HTML settings file. This data defines
the available languages available for the speech API.
These language tags are used to filter SpeechSynthesis' API `getVoices`
result. Tags must use BCP 47 format.
https://developer.mozilla.org/en-US/docs/Web/API/SpeechSynthesisVoice/lang
Add a server-side app for the audio captions feature and record proto-events
for this data.
As it is, only behaves as a pass-through module. The idea is to include all
the business intelligence in this app.
There's a VoiceUser cleanup procedure bound to the User's cleanup
routine in Meteor's server-side. That cleanup is _silent_ and does not
use a dedicated modifier from voice-user et al, which is not
straightforward and might waste a few minutes of understanding what's
happening when debugging audio collections.
This commit centralizes that cleanup in a new clearVoiceUser modifier in
voice-user as well as logs when it works.
Sometimes the handler that listens for the state change in the callState is
not updated correctly.
In these rare cases, the state of the callstate changes directly to in_conference,
not taking the expected path: call_started -> in_echo_test -> in_conference
Fixes a case when the presentation is just uploaded and a wrong initial zoom was set.
Also fix viewer zoom not correclty adjusting to the area size when zoomed out.
There are scenarios where the full audio broker (SFU) stop procedure
may be called multiple times in a very short timestamp - eg a concurrent
stop + connection failure; a timeout in the transfer procedure + a
reconnect attempt, [...]. When that happens, calls to exitAudio may throw
errors if the broker was already released - and that's not the expected
behavior.
If a viewer session failed mid-call, it was being scheduled for a reconnect via
the min-max connection timers (30s-60s), which is terrible UX.
This commit makes screen sharing viewers try to reconnect immediately when
appropriate.
Outbound/presenter screen sharing reconnect was broken from inception, so it's
being removed until it´s properly re-implemented.
This also fixes an issue where presenter disconnections would be silent for the
end user - now an error toast is shown and the error properly logged.
Fixes an issue where subsequent failures might lead to wrong error codes being
reported;
Splits the screen sharing bridge stop method into a reconnect-safe version and
a public one - should also address some quirks with inbound stream reconnection.
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.
There could be a scenario where the local gDM stream wasnt cleaned up;
eg.: SFU is offline.
This commit guarantees all tracks from the local stream are stopped.
FreeSWITCH has mDNS resolution capabilities as of 1.10.7. Having the filtering
configurable in the client allows us to field trial whether we should keep that
on or off. The default is still to filter them out because FreeSWITCH does not
resolve mDNS candidates by default (ice_resolve_candidate in switch.conf.xml).
- 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
Adds support for multiple cameras pins.
The pinned cameras are stored in a FIFO-type queue
When a camera is pinned the oldest one is removed.
The queue size can be set via create parameter 'maxPinnedCameras',
if not defaults to 3.
