When the voice-limit guard (or any JWT service) answers 403 with a reason —
"This voice channel is full.", "You don't have permission to …" — EC wrapped it
in FailToGetOpenIdToken and the user saw the generic error page with
OPEN_ID_ERROR. A 403 MatrixError with a non-empty `error` now throws
SFUTokenRefusedError ("Can't join this call" + the server's sentence), so the
reason is the description. Other failures are unchanged. Unit-tested.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
"Mute Screenshare Audio" (io.lotus.set_deafen screenshareAudioMuted) used
RemoteParticipant.setVolume(0, ScreenShareAudio). EC's own createVolumeControls
writes volume 1 through the same setter the moment a new screenshare media item
resolves, so when the sharer stopped and re-shared (or a late joiner shared)
the audio came back at full volume while the host button still said
"Unmute Screenshare Audio". Reproduced on the local calls stack with two
headless clients: after a re-share the screen_share_audio element read vol=1.
Now the flag is a global behavior (muteScreenshareAudio$) that
LivekitRoomAudioRenderer turns into the `muted` prop of every
Track.Source.ScreenShareAudio element — the exact mechanism deafen already uses
(pub.setEnabled(false): the server stops sending). Verified via the
RemoteTrackPublication behind each <audio>: the re-published track (new sid)
mounts with enabled=false while muted and re-enables on unmute; deafen +
undeafen leaves it muted; teardown resets the flag so the next call starts
clean. Unit tests updated; renderer test asserts only ScreenShareAudio
elements get muted by the new prop.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01PPmy3tPq869XDW4njjVaKA
* make `delayed_leave_event_delay_ms` and `network_error_retry_ms` mandatory
* Support delegation for legacy jwt request
* Calculate `maximumNetworkErrorRetryCount` based on timeouts
Add a `media_quality` section to config.json that allows self-hosters
to configure video codec, resolution, bitrate, framerate, and simulcast
layers for both camera and screen sharing.
This addresses the long-standing request in #249 for configurable media
quality settings. The LiveKit SDK already supports all of these options;
this change exposes them through the existing config system.
New config.json fields:
- media_quality.video_codec: preferred codec (vp8/vp9/h264/av1)
- media_quality.video: camera resolution, bitrate, framerate, simulcast layers
- media_quality.screen_share: screen share resolution, bitrate, framerate,
simulcast layers (enables 3+ layer simulcast for screen sharing)
All fields are optional and fall back to the existing defaults (VP8,
720p camera, 1080p screen share) when not specified.
Signed-off-by: Ryan Emmick <ryanemmick4@gmail.com>
- use the new domain logic to discover the transport
- then try to authenticate
- Also fix the bug in multi sfu where active$ not updated on delayId change
once)
The local jwt token needs to be aquired via the right endpoint. The
endpoint defines how our rtcBackendIdentity is computed. Based on us
using sticky events or state events we also need to use the right
endpoint. This cannot be done generically in the connection manager. The
jwt token now is computed in the localTransport and the resolved sfu
config is passed to the connection manager.
Add JWT endpoint version and SFU config support Pin matrix-js-sdk to a
specific commit and update dev auth image tag. Propagate SFU config and
JWT endpoint choice through local transport, ConnectionManager and
Connection; add JwtEndpointVersion enum and LocalTransportWithSFUConfig
type. Add NO_MATRIX_2 auth error and locale string, thread
rtcBackendIdentity through UI props, and include related test, CSS and
minor imports updates
Lotus CIandClaude Opus 5
Ryan Emmick
Timo K.
Robin
Valere
fkwp
Timo
Will Hunt