SubscriptionManager tracks data-track subscriptions but exposed no accessor
for the active ones, so callers could see subscribed and published media tracks
and published data tracks, but not subscribed data tracks. Add
GetSubscribedDataTracks(), returning the bound data down-tracks, mirroring
GetSubscribedTracks() for media, so a participant's full track set can be
accounted for. Regenerated the LocalParticipant fake.
Co-authored-by: Claude Opus 4.8 <noreply@anthropic.com>
* report a reason on room-ended telemetry
Room.Close already took a ParticipantCloseReason, but OnClose was func()
with no arguments, so the reason was structurally dropped before it could
reach RoomEnded.
Add types.RoomCloseReason and carry it through Close -> OnClose -> RoomEnded,
where it lands on both the analytics event and the room_finished webhook.
Room.Close now takes the room reason and derives the participant reason from
it, so the two can never disagree. ToParticipantCloseReason maps each reason
to exactly what its call site passed before, so no participant-facing
behaviour changes; a table test pins that mapping.
* fix(test): update webhook test for RoomCloseReason
test/ was outside the packages checked before pushing, so this Close call
site was missed. Also assert the reason reaches the room_finished webhook,
which the unit test cannot cover since it runs with a nil notifier.
* Reduce locking in media track + telemetry listener on move participant.
Telemetry listener was not getting it from new room on room move.
* update comment
* fix signal bytes reporting
* room aware telemetry listener in media track
* data track telemetry listener
* Check for ICE connection before closing participant on signal close.
Only close the participant if ICE has not connected by the time signal
source is closed. If ICE had connected, candidates have been exchanged
and link was established. So, it should be resumable.
Waiting for DTLS closed the participant sometimes in the windowa after
ICE connection, but before DTLS finishes and that unnecessarily closed
the participant forcing a full reconnect.
* variable name
* Add configurable read-message size limit on signalling WebSockets
Set a read limit on both the client-facing (/rtc) and agent worker
WebSocket connections so an oversized frame is rejected by the transport
before being buffered. The limits are operator-tunable via
signal_message_size_limit and agent_signal_message_size_limit, both
defaulting to 2 MiB (0 disables).
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
* Add tests for signalling WebSocket read-message size limit
Cover the configurable signal_message_size_limit added in the prior
commit:
- config: assert both limits default to 2 MiB and that a YAML override
(including 0 to disable) is parsed correctly.
- full-path integration: a real client connects to /rtc on a single-node
server and an oversized frame is rejected by the transport with a 1009
close; a 0 limit leaves the connection unbounded and signalling
proceeds.
Adds setupSingleNodeTestWithConfig so a single-node server can be started
with config overrides.
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
* Bound decompressed size of signalling WebSocket messages
conn.SetReadLimit only accounts for the compressed bytes read off the
wire, and the client-facing /rtc upgrader negotiates permessage-deflate,
so a small compressed frame could still expand into a much larger buffer
once inflated. Enforce the same limit on the decompressed message by
reading through NextReader + io.LimitReader in WSSignalConnection instead
of the unbounded ReadMessage.
The transport-level SetReadLimit is kept as the cheap wire-level guard;
the new check is the decompressed-size backstop.
Adds NextReader to the WebsocketClient interface (regenerated fake) and a
unit test plus permessage-deflate integration tests covering the
amplification case.
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
---------
Co-authored-by: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
* Fix publish track count on migration in.
https://github.com/livekit/livekit/pull/4707 addressed the case of
publish tracks overcounting due to synthesised track publish on migrate
in. But, it introduced an issue where published tracks count could go
negative because unpublish subtracted the counter irrespective of the
track actually migrated in or not.
Fix it by keeping track of local publish.
Also, the older code was skipping publisher track count increase if the
synthesised publish was handled first. Address it by checking if the
track is actually new (i. e. fresh local publish) when the track was
already created in the migrate in path.
* fix pub time for tracks published after migration
* test
* prevent multiple track egresses
- Count a publish attempt on a migrating in tarck as there is no
AddTrack for that.
- Add cancel publish only if the participant connection is canceled
- Do not add publish counter for synthetic publish attempts which
happens for migrating in tracks. It will be counted on migrating in
node when the track is actually published, i. e. negotiated/packets
flowing.
* Record subscribe stream start time in prometheus.
Adjust for mutes, i. e. take the last unmute time as the start point and
calculate time till the first byte is sent.
* close the tiny window of race
* Prevent long tail sample when publisher glitches.
Thanks to @milos-lk for this.
Publisher restarting would have reset the layer and would have caused a
sample with very high stream start time. We only need to capture when we
do a dummy start or when the state is seeded to a different node upon
migration.
* reduce a diff
* test
* changed the wrong thing, thank you Devin
* Async attributes on participant.
How it is different from existing participant attributes?
1. Async attribute can be added one at a time.
2. These are not included in `ParticipantInfo`.
3. Get an attribute bt participant identity and async attribute ID as
and when needed.
* clean up
* get full definitions, not just ids
* listener OnDataTrackSchema
* name length config
* data blob
* deps
* static check
* Add missing request ID
* Update protocol commit
* Wire up StoreDataBlobResponse
* Pass request ID through in GetDataBlobResponse
* deps
* atomic
* sctp at 1.9.5
* remove proto clone
---------
Co-authored-by: Jacob Gelman <3182119+ladvoc@users.noreply.github.com>
* feat: acquire requested video layer directly at HIGH quality by default
Two changes that together remove the visible low->high quality ramp for a new
subscriber (both publisher-first and subscriber-first join orders):
1. Default a subscriber's initial video quality to HIGH on bind instead of LOW
for adaptive stream, so the subscribed max layer is the top layer. Adaptive
stream clients can still scale down afterwards based on viewport.
2. On initial layer acquisition the forwarder/selector latch directly onto the
allocator's target (the requested top layer) instead of opportunistically
latching onto the first lower key frame that arrives. A short
initial-acquisition grace aims the target at the requested layer; if it does
not show up in time, the target falls back to the highest layer seen so
acquisition never stalls.
Always on - no configuration flag.
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
* feat: gate start-at-desired-quality behind EnableStartAtDesiredQuality flag
Put the "acquire requested video layer directly at HIGH quality" behavior
behind a per-subscriber EnableStartAtDesiredQuality flag (default off, so
the original low->high ramp-up is restored unless enabled).
Plumbed from config.RTC.EnableStartAtDesiredQuality through ParticipantParams
-> SubscribedTrack/DownTrack -> Forwarder -> simulcast selector, gating all
three behavior changes: the HIGH default on bind, the forwarder's
initial-acquisition grace, and the selector's direct-latch-onto-target.
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
* Potential fix for pull request finding
Co-authored-by: Copilot Autofix powered by AI <175728472+Copilot@users.noreply.github.com>
* remove config.
---------
Co-authored-by: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
Co-authored-by: Copilot Autofix powered by AI <175728472+Copilot@users.noreply.github.com>
Shared helper for callers that need to distinguish intentional/expected
participant closures (client leave, admin action, room teardown, migration)
from connection failures. Extracted from cloud's IsClosedIntentionally
switch so cloud-side code paths can share a single source of truth.
* rtc: report participant kind code and details
Plumb ParticipantKind and KindDetails through MediaTrack and
BytesTrackStats so track-level reporting can record the numeric kind
code plus details codes on every participant_session aggregation,
alongside the existing Kind string. Also picks up the new kind fields
on resolved BytesSignalStats participants.
Adds deployment/agentID/version to the agent worker logger.
Had made a change in remote participant case to not have telemetry
listener as telemetry does not apply to remote participant. But, that
listener ended up getting used for subscriber and became a null
listener. Use the listener of the subscriber participant for subscribed
tracks.
Release to Docker / docker (push) Failing after 3m42s
* Key telemetry stats work using combination of roomID, participantID
With forwarded participant, the same participantID can existing in two
rooms.
NOTE: This does not yet allow a participant session to report its
events/track stats into multiple rooms. That would require regitering
multiple listeners (from rooms a participant is forwarded to).
* missed file
* data channel stats
* PR comments + pass in room name so that telemetry events have proper room name also
* Make data message naming a bit more consistent.
OnDataPacket and OnDataMessage half-way in the chain made it confusing
(for me at least). Use same name throughout.
API still uses SendDataPacket, but that is not harder to read.
* test
* Use published track for model access in data down track.
No need to pass in the model.
* implementation type assertion
* remove redundant log key
* one more type
* Close interface method
* clean up redundant log key
* remove unused method
* some interface changes to support data tracks in replay
* correct mock signature
* switch participant callbacks to room to listener interface
* mage generate
* clean up
* clear listener
* clean up
* use interface in up data track manager
* tweaks
* Paul feedback - should reduce the diff as this keeps the room handlers as is except making methods for a couple of anonymous handlers
* clean up
* WIP
* check using protocol version
* revert
* clean up
* sdp cid argument
* WIP
* WIP
* test
* clean up
* clean up
* fixes
* clean up
* clean up
* clean up
* conditional checks
* tests for both dual and single peer connection
* test
* test
* test
* type check
* test
* todo
* munges
* combined config
* populate mid
* limit to receive only
* clean up
* clean up
* clean up
* older test
* clean up
* alternative audio codec
* dtx
* don't need to copy
* Anunay feedback
* use the available peer connection
* publisher check
* WIP
* WIP
* WIP
* no mid
* media sections requirement
* mage generate
* WIP
* WIP
* set data channel receive size for test
* handle early media better
* WIP
* do not do ICERestart if no subscriber
* WIP
* WIP
* WIP
* WIP
* WIP
* WIP
* start up subscriber RTCP worker
* WIP
* WIP
* clean up
* clean up
* flag to indicate use of single peer connection
* remove unused interface method
* clean up
* clean up
* Jie feedback #1
* deps
* do not access subscriber in one shot mode
* more places for one shot mode
* more one shot fixes
* deps
* deps
* test
* Populate SDP cid in track info when available.
- Adding SDP cid to TrackInfo. Browsers like FF uses a different stream
id for AddTrack and actual SDP offer. So, have to look up using both
on server side. To make it easier, store both (only if different) in
TrackInfo.
- Use a codec in TrackInfo for audio also. There is some discussion
around doing simulcast codec for audio so that something like PSTN can
use G.711 without any transcoding. So, just keep it consistent between
audio and video.
- Populate SDP cid when SDP offer is received. It could populate a
pending track or an already published track if the new offer is for a
back up codec where the primary codec is already published.
- Passed around parsed offer to more places to avoid parsing multiple
times.
- Clean up MediaTrack interface a bit and remove unneeded methods.
* WIP
* WIP
* deps
* stream allocator mime aware
* clean up
* populate SDP cid before munging
* interface methods
* More v2 signalling changes
- Use wire message in relay signal
- Generate offerId for SDP for sync mode also
- more checks for SDP offer/answer state to see if there are cases of
missed/duplicate messages
* clean up
* HTTP PATCH on participant sid
* rename
* signal handler
* new files
* clean up
* clean up reassembly
* handle messages on server side
* deps
* staticcheck
* clean up
