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.
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.
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.
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.
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.
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.
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
- 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
"default" is not an universally valid default value for deviceIds which was causing issues with Firefox and Safari in some specific scenarios where exact deviceId constraints were being used
Seems to have been introduced by a partial merge commit
There were a bunch of style changes introduced by that partial commit as well; I kept those changes to avoid introducing further conflicts between v2.4-2.5...
When joining breakouts, we now wait for the bridge to be loaded before
automatically start user's audio.
This problems happens only on fullaudio bridge
Restored the old behavior when ending breakout rooms while user is in the
breakout audio transfer, which is to the trigger the reconnection to the audio
in the main room.
This behavior could be improved by (instead of reconnecting) transfering user
back to the main room, but this requires some changes in akka-apps/fsesl
which can be treated in a different issue.
Closes#13242
For browsers that don't support headerBytesSent in RTCOutboundRtpStreamStats
neither headerBytesReceived in RTCInboundRtpStreamStats, we are now able
to calculate upload and download rates.
We are also able to get transportStats information for browsers that
don't support iceTransport attribute of RTCDtlsTransport.
Added support for getStats in screenshare's service. This works similar
to the getStats for video provider, and the information retrieved from
screenshare is added to the video information for cameras.
Here's what we do when user activates mic:
1 - When we do something similar to listenonly's joining process
until we find a valid candidate-pair. The information about this
local candidate is store.
2 - We then start a new userAgent, and as soon as browser finds
a candidate with the same local ip address, we leave only this
candidate in the SDP and send this to FreeSWITCH. SDP should
contain only a single candidate.
3 - The rest of signaling process is basically the same.
This commit contains three fixes: one already reported and two detected
during the investigation of the solution.
This started as a fix for firefox (#12023), but i also fixed the muted
alert/banner when device changes: the banner wasn't detecting device changes,
unless audio was deactived/actived.
There's another fix for the microphone stream: we now keep sender's track
disabled if it was already disabled for the sender's track of the previous
selected device.
Also did small refactor for eslint checking.
Some technical information: in sip bridge (bridge/sip.js), setInputStream and
liveChangeInputDevice function were both fully turned into promises, which
guarantees we have everything ready when it resolves to the respective values.
This helps AudioManager (audio-manager/index.js) to sequentially sets and
tracks the state of the current microphone stream (inputStream), when calling
liveChangeInputDevice function: we first set the current stream to null,
creats a new one and then set it to the newly created value - this is needed
because MutedAlert (muted-alert/component.jsx) can then gracefully
allocate/deallocate the cloned stream when it is set to a non-null/null value
(the cloned stream is used for speech detection with hark).
In MutedAlert we also make sure to enable the cloned stream's audio
tracks, just in case the user change the device when muted (audio track is
disabled in this case), which also leaves the cloned stream muted (we then
enable the track to allow speech detection).
Closes#12023
