Mostly benign, but exitAudio/forceExitAudio was throwing an unhandled
error when called on sessions with no active audio because the
underlying bridge methods did not check whether there was an active
session to stop beforehand.
Several improvements to tldraw whiteboard:
- Only send the shape diff on shape updates (reduce a lot the message traffic)
- Shape permissions (don't allow others to select/edit unless you are presenter/moderator)
- This required some changes in akka model
- Tldraw state patch changes to improve stability with fast updates (fix several crashes)
We were calling upsert in the Annotations collection for the same annotation in all frontend instances, this could lead to the same annotation being inserted
multiple times with different ids due to concurrency.
Added the html5InstanceId of the original request to the redis message so we can use it to only call upsert in one instance.
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.
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).
