* test: stop registering the global DefaultConfig with the logger
InitLoggerFromConfig hands the pointer to zaputil.ComponentLeveler as its
level resolver, and protocol a879e94 gave logger.Config a
ResolveComponentLevel that takes c.lock. The two test packages passed
&config.DefaultConfig.Logging, so every first-time component level
resolution locked a mutex inside the global DefaultConfig while
config.NewConfig marshalled that same global from another goroutine:
Read at 0x39baa08 yaml.Marshal -> pkg/config/config.go:633 (NewConfig)
Write at 0x39baa08 ComponentLeveler.resolve -> zaputil/leveler.go:117
That tripped the race detector in pkg/rtc TestPreferMediaCodecForPublisher
and broke CI on master.
Register a test-local LoggingConfig instead, so the logger never touches
the global that NewConfig reads. DefaultConfig.Logging only ever sets
PionLevel, so behavior is unchanged. A plain struct copy is not an option
here: logger.Config holds a mutex and vet's copylocks would reject it.
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
* test: make RTCClient callbacks safe to set while the client runs
The scenario tests assign c2.OnDataReceived after waitUntilConnected, so
the assignment raced the data channel goroutine already reading the field
in handleDataMessage:
Read test/client.(*RTCClient).handleDataMessage client.go:1146
Write test.scenarioDataPublish.func1 scenarios.go:160
This is what failed TestMultinodeDataPublishing on master.
Replace the three exported callback fields with atomic pointers behind
SetOnConnected/SetOnDataReceived/SetOnDataUnlabeledReceived. OnConnected
and OnDataUnlabeledReceived have no writers today, but they are read from
the same background goroutines and would race the moment one appeared, so
all three move together rather than leaving a split API on one struct.
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
---------
Co-authored-by: Claude Opus 5 (1M context) <noreply@anthropic.com>
pion no longer starts the repair stream reader when a custom BufferFactory
is set, so the mid/rid/rsid extensions were never observed and simulcast RTX
streams were never paired with their primary streams.
Extract the extensions on the buffer write path instead. Migrated publishers
send no extensions at all, so pair those from SimTracks.
Adds an integration test covering both paths, and moves the vnet setup it
shares with the downtrack test into pkg/testutils/vnettest.
Co-authored-by: Claude Opus 5 <noreply@anthropic.com>
* 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>
* 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>
* 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
* Make new path for signalling v1.5 support.
To be able to support newer clients to interact with older servers, move
signalling v1.5 to new path (`/rtc1`). On the new path, `join_request`
is required and single peer connection is used.
With the existing path `/rtc`, single peer connection is still supported
if `join_request` is used.
Newer clients connecting to old server should follow
1. Try new path WebSocket
2. If that fails, try new path validate at `/rtc1/validate`.
3. If the above gets a 404 which will happen with older server, revert
back to old path and signalling 1.0.
Open to suggestions on path name.
* test on both paths
* change path from /rtc1 -> /rtc/v1
* test all rtc service path combinations
* 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
* Send error response when update metadata fails.
Keeping it simple for the first implementation.
- Send error response only if request_id != 0
- Two kinds of errors notified
o does not have permissions - NOT_ALLOWED
o attributes exceeds size limits - INVALID_ARGUMENT
* Check size limits on metadata and name set from client.
Added a name length limit also.
* check name length in service update participant path also
* limit check in limit config
* update protocol
* longer keys
* Support for participant attributes
* move metadata setters to LocalParticipant
* address feedback
* forward error
* update go mod
* update attributes first
* Integrate logger components
Dividing into the following components
* pub - publisher
* pub.sfu
* sub - subscriber
* transport
* transport.pion
* transport.cc
* api
* webhook
* update go modules
* Avoid reconnect loop for unsupported downtrack
If the client subscribes to a track which codec is unsupported by the
client, sfu will trigger negotiation failed and issue a full reconnect
after received client answer. If the client try to subscribe that track
then it will got full reconnect again. That will cause a infinite
reconnect loop until the client don't subscribe that track. This PR
will unsubscribe the error track for the client and send a
SubscriptionResponse that contain the reason to indicates the track's
codec is not supported to avoid the reconnect loop.
* WIP commit
* SubscribedQualityUpdate message to send list of currently subscribed
qualities for a simulcast video publisher
* Correct subscriberID
* goimports
* Do quality update on add/remove of subscribed track
* do not update quality when admin mute is active
* update quality on admin unmute
* Update protocol version
* Simplify max subscribed quality loop per David's suggestion
* Stream Allocator Try 3
Making an intermediate PR to do
- Special treatment for screen share tracks
- When allocating all tracks,
o try to stream all tracks by starting with the lowest layer
o multi-pass across tracks to get a more even distribution
Not yet done:
-------------
In deficient state,
o Allocate a specific track on a change
o Steal from other tracks
* Correct sense of managed track
* have to range to copy
* generate
* fix VideoLayers compare
* Use t.simulcasted
* Allow participants to create new connection for publishing new tracks
new param 'publish=<name>' enables publishing new connection with publish
only grants. Also, identity is appended with '#<name>'
* Add integration test to duplicate publish only connection
Signed-off-by: shishir gowda <shishir@livekit.io>
* Reuse token in test and set subscribeGrant to false on publish param
Signed-off-by: shishir gowda <shishir@livekit.io>
* create auth token with grants
Signed-off-by: shishir gowda <shishir@livekit.io>
* Signal ice on negotiate in test client
Signed-off-by: shishir gowda <shishir@livekit.io>
* Update pkg/service/rtcservice.go
Co-authored-by: David Zhao <david@davidzhao.com>
Co-authored-by: David Zhao <david@davidzhao.com>
* prestop and hasparticipants in interface
* add prestop function to existing routers
* fakerouter prestop
* update protocol version
* read lock
* redis router graceful stop
* test fix
* force stop
