* service: start room composite egress configured in a room preset
StandardRoomAllocator.CreateRoom merged a room preset into a copy of the
request and discarded it. RoomInternal has no field for room composite
egress, and getOrCreateRoom started it from the caller's original request,
so a room composite egress from a preset was never started.
RoomAllocator.CreateRoom now also returns the request with the preset
applied, and getOrCreateRoom starts the egress from it.
* service: return ErrEgressNotConnected instead of calling a nil egress launcher
getOrCreateRoom started a room composite egress without checking that an
egress launcher exists. RoomService.CreateRoom rejects egress on the request
when there is none, but not egress from a room preset or a token's roomConfig.
* Increase receiver loadbalance threshold when batch io enabled
With batch IO the send syscall moves to the BatchConn flush goroutine, so WriteRTP only
translates the packet and enqueues it. Increash the threshold to reduce
* lint
* 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.
* fix: bound data-track buffering under downlink congestion
The SFU data-track down-channel writer was constructed with targetLatency=0,
which disables the buffered-amount/latency-based drop in writeUnreliable. Under
subscriber downlink congestion, frames were queued unbounded and in order in the
per-subscriber SCTP send buffer instead of being dropped, so end-to-end latency
grew without limit and only drained once congestion cleared. This contradicts
data tracks being a low-latency, lossy transport.
Wire a dedicated `datachannel_data_track_target_latency` config through to both
data-track writer call sites (mirroring how `datachannel_lossy_target_latency`
controls the lossy publishData channel), reusing lossyDataChannelMinBufferedAmount
as the drop floor. Defaults to 100ms so data tracks are latency-bounded out of
the box; set to 0 to restore the previous unbounded behavior.
Fixes#4666
* fix: track drained bytes on successful unreliable data channel writes
writeUnreliable only called BitrateCalculator.AddBytes when a write failed,
so on the common success path the calculator never received samples. As a
result Bitrate() had no data and the latency-based drop threshold collapsed to
the static minBufferedAmount floor, making the per-subscriber latency control
(datachannel_lossy_target_latency and the new datachannel_data_track_target_latency)
effectively inert. Call AddBytes after every write, matching writeReliable.
* 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>
* service: enforce metadata size limit in CreateRoom, bump default to 512 KiB
CreateRoom previously accepted any metadata size; only UpdateRoomMetadata
rejected oversized payloads. Mirror the same CheckMetadataSize check at
the CreateRoom API boundary so both entrypoints are bounded.
Default MaxMetadataSize moves from 64000 to 512 * 1024 to match the
practical needs of customers using room metadata for richer state. The
limit remains configurable via the existing limits.max_metadata_size knob.
* service: split room vs. participant metadata limit, enforce on join + agent dispatch
LimitConfig.MaxMetadataSize was shared between room metadata and
participant metadata. Last commit's bump to 512 KiB lifted both ceilings;
this restores the participant ceiling to 64 KB and introduces a separate
MaxRoomMetadataSize (default 512 KiB) for room metadata.
Additional enforcement:
- RoomManager.StartSession rejects joins whose JWT-grants metadata or
attributes exceed the participant/attributes limits. The check was
missing entirely from this path.
- AgentDispatchService.CreateDispatch and the embedded
CreateRoomRequest.Agents path now validate metadata and attributes
against the common 64 KB ceilings (previously unbounded).
NewAgentDispatchService gains a LimitConfig parameter; the two wire_gen
callsites are updated.
* service: collapse metadata size limit to single 512 KiB knob
Reverts the LimitConfig split introduced in the previous commit:
MaxRoomMetadataSize, CheckRoomMetadataSize, and the max_room_metadata_size
yaml key are removed. MaxMetadataSize moves back to 512 * 1024 and gates
all metadata uniformly — room (CreateRoom, UpdateRoomMetadata), participant
(UpdateParticipant, signal UpdateMetadata, JWT grants on join), and agent
dispatch (CreateDispatch + embedded RoomAgentDispatch).
MaxAttributesSize stays at 64 KB and continues to gate participant and
agent-dispatch attributes separately.
Test cases consolidated under the single knob.
* kb -> kib
There are several places the participant can drop off after initiating a
connection attempt. Count those places as cancellation including when
participant is closed due to specific reasons.
Cancels should be discounted when determining RTC/ICE connectivity
success/failure percentage.
Sfu will fallback to retransmit packet by media stream ssrc if rtx
is not negotiated (client doesn't have), so we should not disable
rtx explicitly (by codec config).
Fix#4519
* Add TURN permission handler.
- Turn off permissions to private/link local/multicast and internal IPs
- Add a list of CIDRs that can be used for more things to deny
permission to.
* unused
* add config for allowing private IPs, used in testing
* add a TTL to user name and use it to auth
* allow list for restricted peer CIDRs
`iceServersForParticipant` builds UDP TURN URLs by interpolating the
node IP directly into a format string:
fmt.Sprintf("turn:%s:%d?transport=udp", ip, port)
When `NodeIP.V6` is set, `ToStringSlice()` includes the bare IPv6
address, producing URLs like:
turn:2a05:d014:ee4:1201:7039:38c:f652:a252:443?transport=udp
RFC 3986 §3.2.2 requires IPv6 addresses in URIs to be enclosed in
square brackets. Without them the port is ambiguous and WebRTC clients
(e.g. libdatachannel) reject the URL with "Invalid ICE server port".
Use `net.JoinHostPort` which handles bracketing for IPv6 and is a
no-op for IPv4, producing well-formed URLs:
turn:[2a05:d014:ee4:1201:7039:38c:f652:a252]:443?transport=udp
turn:1.2.3.4:443?transport=udp
* fix: ensure num_participants is accurate in webhook events (#4265)
Three fixes for stale/incorrect num_participants in webhook payloads:
1. Move participant map insertion before MarkDirty in join path so
updateProto() counts the new participant.
2. Use fresh room.ToProto() for participant_joined webhook instead of
a stale snapshot captured at session start.
3. Remove direct NumParticipants-- in leave path (inconsistent with
updateProto's IsDependent check), force immediate proto update,
and wait for completion before triggering onClose callbacks.
* fix: use ToProtoConsistent for webhook events instead of forcing immediate updates
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
* Add support for RTP stream restart.
When an unhandled packet is encountered, try a restart sequence.
Restart happens when 5 packets with contiguous sequence numbers and same
or increasing time stamps are received. Note that this does not work for
B-frame type of scenarios, but that is true for receive path handling
even before this. As WebRTC does not use B-frames, it is fine. But,
needs to be looked at again if B-frames are necessary.
It is controlled by a config that is disabled by default.
* clean up
* debug log
* 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
- Move downTrack instantiation to SubscribedTrack as it should own that
DownTrack. Still more to do here as `DownTrack` is fetched from
`SubscribedTrack` in a few places and used. Would like to avoid that,
but doing this initially.
- Use an interface from sfu.Downtrack and replace a bunch of callbacks.
SubscribedTrack is the implementation for DownTrackListener.
* 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
* 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
Martin Njalale