There may be other bridges may not need to force relay traffic on
firefox as @prlanzarin pointed out. So set the default to a
configuration that works out of the box but leave other choices for the
operator.
The option is moved from kurento namespace to media next to the general
forceRelay option.
Firefox has a buggy ICE implementation and needs WebRTC media traffic to
be routed through a turn server to work reliably with mediasoup.
Use the information fetched by the STUN API to determine if the operator
has configured a turn server. If there is one force firefox to use it.
Closes#16164
When BBB is run as a single node or in a scaleout setup with a cluster
proxy in front (see https://docs.bigbluebutton.org/admin/clusterproxy.html) it
is useful to store client settings in browser localStorage instead of
sessionStorage. If localStorage is configured then the client will keep seetings
like notifications for user joining, chat etc across meetings.
It is not advisable to set the setting to `local` in a setup of multiple BBB
nodes without a cluster proxy in front of it because this would lead to
unexpected behaviour at users. The browser would store settings for each server
and for users it would look like BBB is sometimes store the settings and
sometimes not.
It adds the new setting
```yaml
public:
app:
userSettingsStorage: (session|local)
```
Currently, collection cleanup code is only run when an added event
is received from the server. Where that fails is in scenarios where
a server-side collection turns empty while an affected users is
disconnected - and then reconnects. There's no removed (or updated)
event so no cleanup code is run and you have stale data.
This commit guarantees a stale data check is run whenever a subscription
is established again. The `added` check was also maintained, although
I'm not too sure anymore it's is still needed. That may need to be
revisited.
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.
BBBVideoStream 'inactive' event currently does not send the
MediaStream id as the payload for handlers. This can cause theoretical
race conditions due to media stream mismatches in places where
'inactive' handling is done globally (eg video-preview).
This adds the MediaStream id to BBBVideoStream's 'inactive' event
and uses it in video-preview to avoid such race conditions.
Extract the deviceId again from the stream to guarantee consistency
between stream DID vs chosen DID. That's necessary in scenarios where,
eg, there's no default/pre-set deviceId ('') and the browser's
default device has been altered by the user (browser default != system's
default).
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
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
