* Flush pending signal responses before closing the web socket.
When the request direction of a signalling connection goes away, the
web socket was closed right away. Responses that the participant had
already sent were dropped.
This loses the leave request on migration. The media node writes
leave(RESUME) and closes the signalling connection just after. The
close won the race, so the client saw a plain web socket close with
code 1000 and never got the leave. It then did a full reconnect
instead of a resume.
Now the response pump is signalled first and drains what is pending,
then the web socket is closed. The producer closes the response
source after its last write, so draining till the source is closed is
a complete flush. A deadline bounds the case where the source stays
open.
The web socket is still closed on all paths, so the ping worker does
not leak.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
* Give the response pump a margin over the drain deadline.
Both waits used the same timeout and started at about the same time.
So when the drain ran to its deadline, the outer wait could give up at
the same moment and close the web socket while the pump was still
writing. That write failed and the message was lost. It also logged a
timeout even though nothing was stuck.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
---------
Co-authored-by: Claude Opus 5 <noreply@anthropic.com>
* Close web socket connections in all paths.
There was a leak of WebSocket pingWorker if the initial response write
errored as it did not close the WebSocket connection.
* graceful close
* 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>
* Prometheus metric for join latency.
Also including a couple of other failures in the signal connection path
and moving the signal connected to after all that.
Not doing counters for the new signal failure paths. I should not have
done for the other two I added a little while ago also (
validation failure and start participant failure) as those are not
scalable to keep adding to node stats. Will probably remove those two
from node stats later. Can add those counters if they are useful.
* deprecate signal failed counters
* Metrics for participant active, i. e. fully established.
- Egress stub for v2 API
- Fix the participant canceled counter 🤦
- Add active counter -> this is increment when a participant becomes
active, i. e. primary peer connection established. Can be used to
monitor node wise connection establishment issues.
- Add singnalling validation fail counter.
With this, we have
- signalling validation fail
- signalling failed --> this is when the `startSession` fails
- signalling connected -> signalling is succesful and can send back
joinResponse to client
on media connection side
- rtc_init -> start
- rtc_connected -> participant session created (joined)
- rtc_active -> primay peer connection established
- rtc_canceled -> could not proceed with RTC connection due to not being
able to resume.
* signalling counters deps
* revert pion/webrtc to 4.2.12 to get SCTP without interleaving
* go back to pion/webrtc 4.2.11 and sctp 1.9.5
When a client hits /rtc/v[01]/validate with a base64 WrappedJoinRequest
whose embedded JoinRequest.ClientInfo is unset, validateInternal called
AugmentClientInfo with a nil *ClientInfo and panicked at ci.Address =
GetClientIP(req). The non-wrapped branch already allocates via
ParseClientInfo; do the same here so pi.Client always gets at least the
resolved client Address.
* Log join duration.
Also revert the "unresolved" init. Defeated the purpose of log resolver
as it was resolving with those values even if not forced. Instead set it
to "unresolved" if not set when forced.
Join duration is not reset if resolver is reset as that happens on
moving a participant and there is no new join duration in that case.
* explode
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 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
* 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>
* Rework node stats a bit.
Related protocol PR - https://github.com/livekit/protocol/pull/1023
- Make a config for node stats measurements. Wanted to put the config in
`routing` package, but a circular dependency forced me to put in
config.go
- Make rate calculations explicit, i. e. requested via config.
Previously, it had some odd checks to decide when to calculate rate
and it would have been calculating over different windows.
- Report signal/data channel bytes every 5 seconds to stats collection
module. Previously, it was doing it every 30 seconds and that meant
some windows could have had a large spike
NOTE: Still need to think about this for load calculations as a large
number of participants leaving could flush in a small window and that
could report a large spike in bytes/packets. Maybe need to ignore
signal bytes for load calculation?
* deps
* use default node stats config if given config is nil
* split out node stats into a struct for re-use
* update config
While working on a different service, noticed delegation of setting up
routes to the service itself. So, making that change and making some
methods internal only as there is no need to export them outside the
service.
* Debug FF TWCC
* - TURN off TWCC for Firefox. Seems to fail with VP9 send, i.e. there are
no TWCC feedback packets when sending VP9.
- Relax thresholds for congestion as staging data is showing
oscillations.
- Clean up some logging.
* debug log a few more signal messages
* revert config
* revert config
* clean up
* Handle room configuration that's set in the grant itself
* ensure refresh token contains updates
* deps
* dep
---------
Co-authored-by: Paul Wells <paulwe@gmail.com>
* Ref count the stats worker.
NOTE: Don't liek this much, but wanted to open this get some 👀 on
this and get feedback.
There are two entities, one for counting signal bytes and another for
media stats. They both send `ParticipantJoined` and `ParticipantLeft`
event.
In the case of a participant resume, as the old web socket
connection is closed, that triggers a signal stats counter close. That
would call `ParticipantLeft` and that would close the stats worker.
The closed stats worker got reaped in `FlushStats` after three minutes.
So, all events after that did not have a worker and hence went
unreported including missing participant_left webhook because it relied
on checking if a participant was ever connected and that needed to check
the worker state.
Using a ref count to keep track of join/leaves. And not close the worker
until ref count goes down to 0.
* create a stats worker on resume
* revert incorrect changes
* transfer connected state
* transfer connected state when creating worker
* resolve participant on a resume
