* fix(users-context): add missing logs
* fix(user-persistent-data): collection publication selector for viewers
Fixes the collection's selector when publishing it to viewers.
* fix(users-context): correctly add user persistent data
Changes the logic of the add_user_persistent_data action in users
context, so that the user information already in the context is merged
with the new one. Also, do not flip the logged out status of users added
by user_persisted_data anymore.
We should be able to capture WebRTC stats in some form for post-processing
so that it helps on debugging support requests (and other use cases, e.g.:
improving field trial analysis on test servers).
Although much of WebRTC stats information can be gathered via server side
components, none have logs as structured for proper post-processing as
the client logs - so we're going the client route for now.
Capture WebRTC stats information for audio and screen sharing via:
- Audio logCodes: new `stats` extraInfo field
- `audio_joined`
- `audio_failure`
- `sfuaudio_error_retry_through_relay`
- `sfuaudio_error_try_to_reconnect`
- Screen share logCodes: new `stats` extraInfo field
- screenshare_presenter_start_success
- screenshare_viewer_start_success
- screenshare_broker_failure
Additionally, add an option to periodically capture WebRTC stats information
for all relevant peers. This is disabled by default since the log can be
verbose (and, consequentially, network taxing when using external
logging targets). It can be enabled via `public.stats.logMediaStats` in
settings.yml. The default interval is 30s. The periodic log format is as
follows:
- logCode: `mediaStats`
- extraInfo.stats: an aggregated stats object of all peers (equivalent
to the `Copy` function in the Connection Status modal).
We currently use full renegotiation for audio, video, and screen sharing
reconnections, which involves re-creating transports and signaling channels
from scratch. While effective in some scenarios, this approach is slow and,
especially with outbound cameras and screen sharing, prone to failures.
To counter that, WebRTC provides a mechanism to restart ICE without needing
to re-create the peer connection. This allows us to avoid full renegotiation
and bypass some server-side signaling limitations. Implementing ICE restart
should make outbound camera/screen sharing reconnections more reliable and
faster.
This commit implements the ICE restart procedure for all WebRTC components,
based on bbb-webrtc-sfu >= v2.15.0-beta.0, which added support for ICE restart
requests. This feature is off by default. To enable it, adjust the following
flags:
- `/etc/bigbluebutton/bbb-webrtc-sfu/production.yml`: `allowIceRestart: true`
- `/etc/bigbluebutton/bbb-html5.yml`: `public.kurento.restartIce`
* Refer to the inline documentation; this can be enabled on the client side
per media type.
* Note: The default max retries for audio is lower than for cameras/screen
sharing (1 vs 3). This is because the full renegotiation process for audio
is more reliable, so ICE restart is attempted first, followed by full
renegotiation if necessary. This approach is less suitable for cameras/
screen sharing, where longer retry periods for ICE restart make sense
since full renegotation there is... iffy.
* Send custom user data to akka apps
* Add user custom data to registered user
* Add user custom data to user join event
* Store user custom data in Redis
* Rename userCustomData to customParameters
* Rename xml tag to userdata
* Demo changes
* Revert "feat(captions): no longer writes in the pad"
This reverts commit a76de8c458.
* feat(transcriptoin): Add config options for the transcription backend
* feat(transcription): Add autodetect option to cc chevron
* feat(transcription): Move transcription options into settings modal
* feat(transcription): Set transcription options via userdata
* fix(transcription): Correct userdata for settings transcription params
* feat(transcriptions): options to auto enable caption button
* feat(transcriptions): Option to hide old CC pad funcionality
* fix(transcription): Fix PR comments
* fix(transcription): Refactor updateTranscript to prevent null user and make it more readable
* feat(transcription): bbb_transcription_provider can be set via userdata
* fix(transcription): Use base10 for parseInt
* fix(transcriptions): Fix CC language divider when using webspeech
* fix(transcriptions): Use a default pad in the settings instead of hardcoding 'en'
We still need to use a language pad such as 'en', but in the future we can better
separate these systems.
* fix(transcription): Add a special permission for automatic transcription updates to the pad and restore old per user updates permission
* feature(transcriptions): Include transcriptions submenu and locales
* chore: bump bbb-transcription-controller to v0.2.0
* fix(transcription): Add missing menu files
* fix(transcription): Fix transcription provider options in settings.yml
* fix: setting password for bbb-transcription-controller
* build: add gladia-proxy.log for transcription-controller
* fix(transcriptions): Remove transcript splitting and floor logic from akka apps
* fix(captions): Show long utterances as split captions, show multiple speaker captions
* chore: bump bbb-transcription-controller to 0.2.1
---------
Co-authored-by: Anton Georgiev <anto.georgiev@gmail.com>
If the autoplay block is triggered in listen only, the connection timer
keeps ticking even if the user correctly accepts the audio play prompt.
That causes an audio re-connect once the timeout expires.
Clear the connection timer if the audio bridge starts with
NotAllowedError as a soft error. For connection purposes, the audio join
procedure worked. The autoplay thing is at the UI/UX level, not WebRTC.
This is an initial, experimental implementation of the feature proposed in
https://github.com/bigbluebutton/bigbluebutton/issues/14021.
The intention is to phase out the explicit listen only mode with two
overarching goals:
- Reduce UX friction and increase familiarity: the existence of a separate
listen only mode is a source of confusion for the majority of users
Reduce average server-side CPU usage while also making it possible for
having full audio-only meetings.
The proof-of-concept works based on the assumption that a "many
concurrent active talkers" scenario is both rare and not useful. With
that in mind, this including two server-side triggers:
- On microphone inactivity (currently mute action that is sustained for
4 seconds, configurable): FreeSWITCH channels are held (which translates
to much lower CPU usage, virtually 0%). Receiving channels are switched,
server side, to a listening mode (SFU, mediasoup).
* This required an extension to mediasoup two allow re-assigning producers
to already established consumers. No re-negotiation is done.
- On microphone activity (currently unmute action, immediate):
FreeSWITCH channels are unheld, listening mode is deactivated and the
mute state is updated accordingly (in this order).
This is *off by default*. It needs to be enabled in two places:
- `/etc/bigbluebutton/bbb-webrtc-sfu/production.yml` ->
`transparentListenOnly: true`
- End users:
* Server wide: `/etc/bigbluebutton/bbb-html5.yml` ->
`public.media.transparentListenOnly: true`
* Per user: `userdata-bbb_transparent_listen_only=true`
The refactoring in 838accf015 incorrectly
replaced the wrong parseMessage function in addBulkGroupChatMsgs.js
This bug is only triggered when the option public.chat.bufferChatInsertsMs != 0.
SFU based audio is missing connection timers, which means the join
procedure can go on indefinitely in a couple of scenarios.
Refactor the connection timers added for re-connections in the SFU audio
bridge and make them valid for the first try as well.
Make 1010 errors (connection timeout) retriable when retryThroughRelay
is enabled.
1007 errors are still a large fraction of our overall audio join error
rate. This usually indicates some sort of firewall block or UDP issues
carrier networks. I can't figure out why some scenarios won't trickle
down to relay candidates though - I'm leaning to scenarios where STUN
packets with USE-CANDIDATE are being mangled/lost along the way or
something else that borks the (already fragile) conn checks for ICE-lite
implementations.
Add a new feature called retryThroughRelay which triggers a retry with
iceTransportPolicy=relay whenever audio fails to join with a 1007 error.
The goal is to force relay usage to try and bypass 1007s scenarios that
still happen.
Disabled by default.
