v2.14.1
---
* fix(screenshare): log presenter/viewer stop on all scenarios
* refactor(screenshare): add presenter data to viewer logs
* refactor(video): add video negotiation and flowing logs
* build(mediasoup): 3.14.9
Prevents changes in the presentation state while sharing media from
updating the presentation last state value. Additionally adds a missing
prop value of generic content state.
Removes the duplicate presentation pile dispatch for external video, as
an identical dispatch runs when the external video component is mounted.
This duplication did not cause any noticeable issues for the user but
resulted in the external video being added to the pile twice.
The plugin loader startup logs aren't following the logger convention,
which makes them hard to work with when post-processing logs.
The appended error message is also not useful since we're logging a
Event variant raw (which either outputs {} or nonsense like { isTrusted:
etc }).
Make the plugin "loaded" and "error" logs adhere to logger conventions.
In the future, the error log could use some tuning - there's no useful
info about root cause here.
- Move askForConfirmationOnLeave into AudioContainer
- Get rid of the unstable useMuteMicrophone hook, which returns a new reference every time the user gets muted/unmuted. Use the stable useToggleVoice hook instead.
The isSharing var is content type agnostic, so it's picking up camera as
content to flag the actions-bar button loading state.
Change the loading flag to track isScreenBroadcasting
(contentType=screen, local || global) and isScreenGloballyBroadcasting
(contentType=screen, global only). Fixes the camera as content false
positive as well as the loading state itself.
When going from "no mic" -> mic via the unmute action, the client isn't
unmuting itself after confirming the change. This is caused by not
waiting the liveChangeInputDevice method (which is a Promise) to be
fully executed before unmounting the AudioSettings modal -- the one
responsible for triggering the unmute. Since it unmounts before the
device is changed, the unmute action will be ignored because the device
is still "listen-only" (no mic).
Properly unmute audio when transitioning from "no mic" -> "mic" via the
unmute trigger by waiting for liveChangeInputDevice to resolve.
Additionally, some general improvements to UI/UX:
- Display the AudioSettings modal title when gUM is on prompt mode
- Add specific subtitles to the AudioSettings modal to 1) warn that no
mic is selected 2) Give a hint that the user can test their devices
- Always honor settings.yml's "initialHearingState" state (whether
local echo feedback should be played by default in AudioSettings)
We are missing a way to select transcription languages in some
scenarios, e.g.: listenOnlyMode=false. The audio settings UI is also not
handling item disposition very well on smaller devices.
This commit does the following to improve those blind spots:
- Add the transcription language selector to it whenever applicable
- Add proper styling to the transcription selector
- Handle small screens by changing the disposition of elements to
portrait mode
- Improve how elements are disposed to a more familiar view: Mic ->
Activity Indicator; Speaker -> Speaker test. This is more in line
with how other platforms do audio configuration/pre flight screens.
The current default setting of muteOnStart=false can lead to performance
issues in larger rooms, especially with the "transparent listen only
mode" and LiveKit, as proven by load testing. In contrast,
muteOnStart=true helps mitigate these issues but complicates the entry
process for smaller meetings, such as 1-on-1 sessions or small classes.
Additionally, the ability to override muteOnStart via the API can create
scalability issues if not managed properly.
Add a new akka-apps flag `voiceConf.muteOnStartThreshold` which acts as
a trigger that forces muteOnStart=true when the number of audio
participants reaches the configured threshold. 0 means no threshold
(disabled).
This trigger overrides any API parameter or static configurations
related to muteOnStart, as well as the client's meeting mute actions.
Pending:
- Remove MeetingStatus.meetingMute state side effects from the client's
"Mute all" and "Mute all except presenter" actions. They should no
longer alter the meeting mute state once this becomes default (just
mute users instead).
FS has an intermittent issue where unmuting a HELD channel sometimes
takes significantly (seconds) longer than usual.
conference <XYZ> unmute <WVU> simply gets stuck with no FS_API response,
which delays the unmute action whenever transparent listen only is
active.
Apparently, unholding the channel PRIOR TO unmuting works around the
issue - at least it could not be reproduced with the scenario at hand.
The unmute API already triggered an unhold in FS internally, which is
the reason why this was not done beforehand. The aforementioned issue is
way worse than an extra "redudant" API call, though.
Always unhold audio channels manually _before_ unmuting.
v2.14.0
---
* feat(mediasoup): add least-loaded worker balancing strategy
* feat(mediasoup): worker transposition (off by default)
* feat(audio): dynamic global audio bridge mechanism
* feat: livekit module, initial implementation
* feat(audio): add signaling support for passive-sendrecv role
* feat(freeswitch): overridable UA string
* feat(audio): muteOnStart detection for conditional dialplans
* feat(audio): mute passive-sendrecv clients on start
* feat(audio): support for mute-and-hold on start
* fix(audio): ignore TLO-incapable clients in hold/unhold metrics
* fix(audio): mute/unmute stuck due to inconsistent hold status
* fix(audio): hold/unhold loop when there are multiple sessions per user
* fix(audio): muteOnStart sessions incorrectly muted on breakout transfers
* fix(audio): header-provided userName incorrectly decoded
* fix(audio): stuck unmute due to borked callerIdNum
* fix(audio): correctly decode user name space chars
* !build(npm): set min Node.js version to >=18.0.0
* build: nodemon@3.1.3
* build: ws@8.17.1
* build(mediasoup): 3.14.8
FreeSWITCH incorrectly generates callerNum headers in its ESL events
when specific, special characters are in place. e.g.:
w_etc_0-bbbID-User;Semi (notice the semicolon) will be generated by
FS as w_etc_0-bbbID-User (everything after the semicolon is ignored).
This breaks callerId comparision for session matching in a few places -
one of the is the unmute/unhold toggle control.
Compare callerNum as prefixes instead of exact strings to match those
scenarios.
This is a temporary fix; we should review callerNum generation in the
future (use Caller-Id-Name in FSESL), as well as stop relying on it
for session matching (use UUIDs and/or client session numbers instead).
Both of these are more involved changes, though.
Whenever both microphone lock settings and transparent listen are
active, users fail to join audio due to an oversight in the audio join
permission procedure. It's denying users from *joining* audio if they're
locked, even though they should only be prevented from *unmuting*
themselves.
Change the permission checks for audio join to allow users even when
they're locked. Use the lock setting to force the muteOnStart flag for
locked users instead.
Alway send the unhold command since it doesn't change flip the state
(contrary to the uuid_hold toggle command).
It's not idempotent, though - so always update the internal hold state
to prevent state mismatches preventing channels from being unheld.
We need the voiceUserId (memberId) in IN_CONFERENCE callState events so
that the SFU can bypass dialplan-level mute states when transferring
between breakout rooms.
Add memberId to VoiceCallStateEvent when applicable. For CHANNEL_STATE
events that do not carry it (RINGING, HANGUP), memberId is an empty
string.
This is an edge case with the uuid_hold command being used through
FSESL or fsapi where holding only works via the uuid_hold <toggle>
subcommand, which may cause the channel to be the opposite of what
we want.
Do not execute if the command is asking for the channel to be HELD
and the channel is already HELD.
There's a routine that's supposed to enforce the channel hold state if
mute and hold states are mismatched. This should only happen in case the
user is unmuted and the channel is held to avoid leaving an user without
inbound or outbound audio by accident. The routine currently checks for
the oppositve scenario as well (muted=true,hold=false), which should not
be enforced because of special scenarios (e.g.: audio-only breakout room
transfers, which have their transparent LO mechanism disabled).
Only enforce hold on muted state mismatch IF muted == false.
There are some scenarios where transparent listen only toggle RPCs are
directed to multiple different sessions in the SFU, which might cause a
hold-unhold loop because they're all anchored to the userId.
Append the callerNum to transparent listen only toggle RPCs so that they
are directed to the appropriate audio sessions in all cases.
This is a rework of the audio join procedure whithout the explict listen
only separation in mind. It's supposed to be used in conjunction with
the transparent listen only feature so that the distinction between
modes is seamless with minimal server-side impact. An abridged list of
changes:
- Let the user pick no input device when joining microphone while
allowing them to set an input device on the fly later on
- Give the user the option to join audio with no input device whenever
we fail to obtain input devices, with the option to try re-enabling
them on the fly later on
- Add the option to open the audio settings modal (echo test et al)
via the in-call device selection chevron
- Rework the SFU audio bridge and its services to support
adding/removing tracks on the fly without renegotiation
- Rework the SFU audio bridge and its services to support a new peer
role called "passive-sendrecv". That role is used by dupled peers
that have no active input source on start, but might have one later
on.
- Remove stale PermissionsOverlay component from the audio modal
- Rework how permission errors are detected using the Permissions API
- Rework the local echo test so that it uses a separate media tag
rather than the remote
- Add new, separate dialplans that mute/hold FreeSWITCH channels on
hold based on UA strings. This is orchestrated server-side via
webrtc-sfu and akka-apps. The basic difference here is that channels
now join in their desired state rather than waiting for client side
observers to sync the state up. It also mitigates transparent listen
only performance edge cases on multiple audio channels joining at
the same time.
The old, decoupled listen only mode is still present in code while we
validate this new approach. To test this, transparentListenOnly
must be enabled and listen only mode must be disable on audio join so
that the user skips straight through microphone join.
Transparent listen only is currently only worth it for meetings with a
number of duplex audio channels larger than a certain value (dependant
on system performance). That is due to the fact that global audio
bridges created for the mechanism also use significant CPU (roughly the
same as an unheld duplex channel), which means it's cost is usually
offset only once there are enough potential channels to be held in a
conference.
This commit adds a new optional feature that introduces some dynamicity
for the mechanism: it'll only be triggered after at least
@voiceConf.transparentListenOnlyThreshold number of muted duplex
audio channels are present in a conference.
The default is 0 (always trigger transparent listen only if the general
mechanism is activated).
Support for bbb-html5's server logger (Winston) was removed in
50d445f026, but configuration for it
remained.
Remove the Winston-based server logger configuration from bbb-html5's
settings.yml. There's no direct alternative since this package hasn't got a
server side component to it - replaced by bbb-graphql-middleware,
bbb-graphql-server and bbb-graphql-actions.
Support for clientLog's "server" target (Meteor) was removed in
50d445f026, but configuration and docs
entries for it remained.
There's no alternative for it, so I'm removing the leftover
configuration and doc entries.
* core: get PR number from artifacts on completed testing workflow
* core: split merge and upload blob-report action into a separate file
* ci: do not remove artifact files
FreeSWITCH sometimes drop frames in very long recordings generating
a raw file smaller than the total time recorded. The other issue is the
occasional drifiting.
This adds option to split audio recording in multiple files by
duration to avoid this problem in setups that are prone to it. Disabled
by default.
Restore the `lint:file` npm script. Rationale outlined in
- 07960af
- https://github.com/bigbluebutton/bigbluebutton/pull/13870
> The lint:fix hook fixes/alters things. The lint run script runs eslint over the whole root directory.
> I just want to check linting offenses on a per-file/directory basis without having files messed with or needing a fancy IDE.
