* rtc: apply simulcast rids from SDP for WHIP (one-shot) publishers
In one-shot signalling mode HandleRemoteDescription set the remote offer
directly and never called OnSetRemoteDescriptionOffer, so
updateRidsFromSDP did not run for WHIP participants. Every simulcast layer
then mapped to spatial layer 0 and all but the first were rejected with
'duplicate layer'.
* rtc: test that a one-shot simulcast offer applies rids to the pending track
* rtc: use h/q rids in one-shot simulcast test, cover single-layer offer
Name the simulcast test layers medium/h and low/q per convention, and add
TestOneShotSingleLayerOfferUnchanged to check a one-shot offer without rids
still yields a single layer with rid "" at spatial layer 0.
* rtc: reclaim abandoned pending tracks
A pending track is freed only when its first media arrives or the
participant leaves. A client that calls AddTrack and then gives up
before any media flows leaves the slot held for the whole session.
After 20 such slots, every later AddTrack is rejected.
Reclaim these slots when the cap is hit, if no media arrived and the
slot is older than the publish timeout. Keep the supervisor monitor in
step for queued tracks that share a track id. Add a publish_timeout
config value, default 30s.
Fixes: #4970
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
* rtc: fix publish_timeout comment to not name an internal constant
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
* rtc: release supervisor-timed-out pending tracks at the limit
Replace the createdAt sweep with the supervisor's publish timeout. The
supervisor counts time only while the publisher is connected and the
track is unmuted, and never reports a live track, so a successor queued
behind a live track is not released.
Restart the publish timer when a queued publish reaches the front of
the queue, so it does not inherit the clock of the track ahead of it.
Write the publication error under the write lock, since it is now read
from another goroutine.
Drop the publish_timeout config value.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* rtc: skip stale supervisor timeouts and keep live monitors
TimedOutPublications re-checks the monitor, because the cached error is
cleared only on the next tick. A track that published since is not
reported, so its queued successor is not released.
RemovePublication keeps a monitor that still holds a published track.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* rtc: drop late media for released pending tracks
Media or an offer with no matching pending track falls back to the first
pending track of the same type. After a release, late media for the
released track could take another pending track's identity.
Record the released track's signal and SDP client ids, the 32 most
recent, and drop media or offers for them. An AddTrack that reuses the
client id clears it.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* rtc: clear all ids of a released track when it is re-added
A re-add cleared only the signal client id, so a Firefox republish that
keeps the old SDP id had its offer and media dropped. Keep the ids of
each released track together, and clear the whole track when a new
AddTrack reuses its signal or simulcast codec client id. The limit now
counts released tracks, and a track ages out with all of its ids.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
---------
Co-authored-by: Claude Opus 4.8 <noreply@anthropic.com>
GetRecentlyDisconnectedParticipants walked the participant update cache
with Get, which moves each entry to the front of the LRU. Walking from
newest to oldest left the cache in reverse order, so after a reconnect
the next updates evicted the most recently updated participants first.
In a room where the subscriber has tracked 128 or more participants, a
participant that had just disconnected was then dropped from the cache
and a later reconnect did not report it as disconnected, and the
out-of-order version check was lost for recently updated participants.
Use Peek, which reads an entry without changing its recency.
selectAlternativeVideoCodec walked the enabled codecs and returned the
first one with a video/ prefix. video/rtx and video/flexfec-03 both have
that prefix, so when the requested codec was not enabled the track could
fall back to RTX. RTX is a retransmission stream, not a real codec, so
the track got registered as video/rtx and subscribers saw a blank tile.
Skip RTX and FEC entries so the fallback is always a valid video codec.
Co-authored-by: Claude Opus 4.8 <noreply@anthropic.com>
* rtc: split EndSession out of MoveToRoom
A participant's session in a room can end without the participant
closing: today on a move, and next on a room re-key, where a room ended
by the sweeper while participants were joining restarts under a new room
id with the same participants in it.
EndSession is the room-independent half of MoveToRoom: report the
published tracks unpublished, report the session end, run the caller's
leave under the ending session's telemetry guard, then take a fresh
guard and reset the deferred resolvers for the next session. The guard
matters: ParticipantLeft releases it and the telemetry service refuses
to open a stats worker for a released guard, so without a fresh one the
next session's join would silently get no worker.
MoveToRoom keeps the subscriber teardown and the onClose fan-out, passed
in as the leave.
Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>
* rtc: rename EndSession to LeaveSession, add JoinSession
EndSession read as terminal, but the participant survives it and joins
another session, so name the pair for what happens to the participant:
LeaveSession and JoinSession.
JoinSession is the other half that review pointed out was missing from
the API: the caller reports the join, then the published tracks are
reported published again under the new session, pairing the unpublish
LeaveSession reported. Without it the re-publish lived only in the
caller, with nothing in OSS proving the two halves balance.
Co-Authored-By: Claude Fable 5.1 <noreply@anthropic.com>
---------
Co-authored-by: Claude Fable 5.1 <noreply@anthropic.com>
participants.
While migrating, the old node will have the recently disconnected
participants in the migrating out participant's cache and needs to sent
out to the participant from the migrating in node via the new WebSocket.
This method can be used to exchange that data between the two nodes.
Keeping it as `recently` disconnected and leaving the semantics upto how
recent for implementation.
* agent: release JobTerminate handler when job ends before registration
JobRequest registered the job's JobTerminate handler after AssignJob
returned, so an ended UpdateJobStatus or worker disconnect handled in
between deregistered nothing and the handler leaked. AssignJob also sent
the assignment before recording the job as running, so an immediate
ended update was dropped.
Record the job before sending the assignment, and after registering the
handler recheck under h.mu that the job is still running and its worker
still registered, deregistering otherwise.
Fixes#4901
* replace utils.CloneProto with proto.CloneOf
* agent: wait on worker readiness and poll for released handlers in test
* fix: hold signal messages until the ReconnectResponse goes out
Clients take the ReconnectResponse as the first message on a resumed or
migrated in signal connection, anything ahead of it is dropped. Hold
messages back until it has been written.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
* fix: keep queued participant updates on a path that always flushes
Queue only while the participant is not ready, that queue is always
drained by the join or reconnect response. Log a dropped SDP, it leaves
the negotiation waiting until the state machine recovers it.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
* fix: flush queued updates whenever the connection opens
Queue participant updates while the handshake is pending again, and give
the signaller a hook that fires when the connection opens, on an explicit
open and on the handshake window expiring, so the queue always drains.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
* fix: drive the handshake window off a timer and read the gate atomically
The window ran only when something asked whether the handshake was
pending, so a connection with nothing else to send held its queue.
Reading the gate under the lock the flush takes closes the race where an
update queued just after a flush.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
* fix: ignore a handshake timeout from a wait that has ended
Stop cannot cancel a timeout that is already running, so tag each wait
and let a timeout act only on its own. Count opens atomically in the
test, the timer fires on its own goroutine.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
---------
Co-authored-by: Claude Opus 5 <noreply@anthropic.com>
SendData API messages do not have sender ID or sequence number. So, they
were not cached and hence excluded from the reliable caching feature
which is meant to provide reliability of data channel messages between
the time other participants see participant as ACTIVE (which happens on
ICE connected) and data channel being open (DTLS done + data channels
opened).
Add a small cache for that and flush those messages on data channel
establishment.
- 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.
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
* 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
* Set publisher codec preferences after setting remote description
Munging SDP prior to setting remote description was becoming problematic
in single peer connection mode. In that mode, it is possible that a
subscribe track m-section is added which sets the fmtp of H.265 to a
value that is different from when that client publishes. That gets
locked in as negotiated codecs when pion processes remote description.
Later when the client publishes H.265, the H.265 does only partial
match. So, if we munge offer and send it to SetRemoteDescription, the
H.265 does only a partial match due to different fmtp line and that gets
put at the end of the list. So, the answer does not enforce the
preferred codec. Changing pion to put partial match up front is more
risky given other projects. So, switch codec preferences to after remote
description is set and directly operate on transceiver which is a better
place to make these changes without munging SDP.
This fixes the case of
- firefox joins first
- Chrome preferring H.265 joining next. This causes a subscribe track
m-section (for firefox's tracks) to be created first. So, the
preferred codec munging was not working. Works after this change.
* clean up
* mage generate
* test
* 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
* HTTP PATCH on participant sid
* rename
* signal handler
* new files
* clean up
* clean up reassembly
* handle messages on server side
* deps
* staticcheck
* clean up
* Add Id to SDP signalling messages.
Allows matching up offer/answer.
For now, the subscriber answer just logs if there is a mismatch.
* correct variable name
* test
* fixing rid
* revert
* clean up
* Add simulcast support for WHIP.
- General change to have rids be anything.
- One issue is rid ordering not matching quality ordering, will need
some dynamic layer quality determination for that.
* clean up
* deps
* test
* Add Moving participant to another room
it is implemented in cloud only since the destination
room can exist in different node with the source room
* Update pkg/service/errors.go
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
* rename
* test panic
* fake LocalParticipantHelper
* revert delete line
---------
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
* Revert "Audio uses signal SignalCid and SdpCid. (#3564)"
This reverts commit cdfbb106d1.
* Revert "Keep simulcast information tied to receiver. (#3563)"
This reverts commit ed5e2f16b2.
* Revert "chore(logs): log VLS type for VP9/AV1 (#3561)"
This reverts commit ad010cfc43.
* Revert "fix(video): determine svc/simulcast from SDP for advanced codecs (#3549)"
This reverts commit 15f565510c.
* chore(deps): update protocol
* Keep simulcast information tied to receiver.
`simulcast` flag in `TrackInfo` is at track lavel. With codec simulcast,
the primary codec (in most cases) is SVC and the backup codec is
simulcast. Back up codec publish changing the track info setting to true
meant that the primary receiver was treated as simulcast if a subscriber
for primary codec joined after the backup codec was published.
Keep track of simulcast flag in receiver.
Also, TrackInfo Cids are from signal. So, keep track of SDP cids
separately. The `simulcastTrackIds` map uses SDP cid. Clean up by all
the SDP cids of a track
* clean up
* clean up
* clean up
* clean up
* test
* Store SdpCid and IsSimulcast in Trackinfo
* clean up
* mock
* Normalize mime type and add utilities.
An attempt to normalize mime type and avoid string compares remembering
to do case insensitive search.
Not the best solution. Open to ideas. But, define our own mime types
(just in case Pion changes things and Pion also does not have red mime
type defined which should be easy to add though) and tried to use it everywhere.
But, as we get a bunch of callbacks and info from Pion, needed conversion in
more places than I anticipated. And also makes it necessary to carry
that cognitive load of what comes from Pion and needing to process it
properly.
* more locations
* test
* Paul feedback
* MimeType type
* more consolidation
* Remove unused
* test
* test
* mime type as int
* use string method
* Pass error details and timeouts. (#3402)
* go mod tidy (#3408)
* Rename CHANGELOG to CHANGELOG.md (#3391)
Enables markdown features in this otherwise already markdown'ish formatted document
* Update config.go to properly process bool env vars (#3382)
Fixes issue https://github.com/livekit/livekit/issues/3381
* fix(deps): update go deps (#3341)
Generated by renovateBot
Co-authored-by: renovate[bot] <29139614+renovate[bot]@users.noreply.github.com>
* Use a Twirp server hook to send API call details to telemetry. (#3401)
* Use a Twirp server hook to send API call details to telemetry.
* mage generate and clean up
* Add project_id
* deps
* - Redact requests
- Do not store responses
- Extract top level fields room_name, room_id, participant_identity,
participant_id, track_id as appropriate
- Store status as int
* deps
* Update pkg/sfu/mime/mimetype.go
* Fix prefer codec test
* handle down track mime changes
---------
Co-authored-by: Denys Smirnov <dennwc@pm.me>
Co-authored-by: Philzen <Philzen@users.noreply.github.com>
Co-authored-by: Pablo Fuente Pérez <pablofuenteperez@gmail.com>
Co-authored-by: renovate[bot] <29139614+renovate[bot]@users.noreply.github.com>
Co-authored-by: Paul Wells <paulwe@gmail.com>
Co-authored-by: cnderrauber <zengjie9004@gmail.com>
* Use new track id for republishing
Always generates new track id for track publishing
to make the behavior more consistent. A republishing of
clien track will be considered as a new track publication
and always trigger trackUnpublished & trackPublished event
on subscriber side.
* remove test
* Negotiate downttrack for subscriber before receiver is ready
This change will save 1 round sdp negotiation time for
subscribing to simulcast-codec or remote node track
* solve comment
* Fix simulcast-codec case
* speed up track publication
Add metrics for track publication and subscription
Return EnabledCodecs in JoinResponse so client can
choose codec without server side codec fallback
Cache remote webrtc track without AddTrackRequest to
let client send publisher offer before AddTrackRequest response
* go mod
* clean code
* Buffer size config for video and audio.
There was only one buffer size in config.
In upstream, config value was used for video.
Audio used a hard coded value of 200 packets.
But, in the down stream sequencer, the config value was used for both
video and audio. So, if video was set up for high bit rate (deep
buffers), audio sequencer ended up using a lot of memory too in
sequencer.
Split config to be able to control that and also not hard code audio.
Another optimisation here would be to not instantiate sequencer unkess
NACK is negotiated.
* deprecate packet_buffer_size
* Disable H.264 for android firefox
* Fix syntax error for rule
* lower case
* Remove disabled codec from AddTrackRequest
* Consistent handling of enabled codecs
Mainly cleaning up where we are doing codec filtering.
There's also behavior change of how we handle codec compatibility. If a client doesn't support the client's desired codec, we'll pick a backup automatically
instead of rejecting the client's request.
Requires an update on multi-codec simulcast handling.
* fix alternative codec selection
---------
Co-authored-by: David Zhao <dz@livekit.io>
Martin Njalale