* turn: accept PROXY protocol on the TCP listener
Behind a TLS-terminating or reverse proxy that dials from its own address,
the embedded TURN server reports the proxy's address to the client as
XOR-MAPPED-ADDRESS. Firefox rejects a loopback or wildcard mapped address
and abandons the allocation, so relay-only clients never get a relay
candidate (#4851).
Add turn.proxy_protocol. When set, the TCP listener requires a PROXY
protocol v1/v2 header on every connection and uses the client address it
carries; connections without the header are rejected. The header is read
before TLS, so it works with both the built-in TLS listener and
external_tls.
* turn: only trust PROXY headers from configured proxies
A PROXY header from any peer that can reach the port would let a direct
client claim an arbitrary source address. Add
turn.proxy_protocol_trusted_cidrs, defaulting to loopback, and close
connections from any other address before reading the header.
getMessageBus built the bus from the redis client alone, so there was no
way to reach the gzip settings psrpc v0.7.6 added at the bus boundary.
Take rpc.PSRPCConfig, which the wire graph already provides, and pass its
bus options to both the redis and the local bus.
Compression is off by default. A peer on an older psrpc cannot decode a
compressed payload and drops it silently, so egress, ingress, SIP and
agent workers all have to be upgraded before quality is raised.
* ingress: add opt-in support for udp:// URL pull ingress
URL pull ingress previously accepted only http, https and srt source
URLs. Add udp:// as an accepted scheme, gated behind a new
`ingress.enable_udp_url_pull` config option that defaults to false, so
unauthenticated UDP sources are not reachable unless the operator opts
in.
Also add a counterfeiter fake for IngressLauncher and a test covering
the scheme validation matrix.
The ingress handler binds a local socket on the caller supplied address
and port instead of connecting out like the http and srt sources do.
Spell out what that means for operators in both the config field doc
comment and config-sample.yaml: caller controlled local port binding,
unauthenticated and spoofable input, and multicast relaying of traffic
on the handler's local network.
Co-authored-by: Claude Opus 5 (1M context) <noreply@anthropic.com>
* config: clarify AdvertiseInternalIP applies to explicit node_ip too
The advertise_internal_ip comment in config-sample.yaml said it works
only when use_external_ip is true. That was accurate until
livekit/mediatransportutil#94, which extended AdvertiseInternalIP to
also be respected on the explicit node_ip path (use_external_ip: false
with rtc.node_ip set) — previously that branch always replaced the
host candidate with node_ip, dropping the original local candidate,
regardless of AdvertiseInternalIP.
Update the comment so it reflects both cases: advertise_internal_ip
keeps the original local candidate alongside the mapped one whether
the mapped IP comes from external-IP discovery or from an explicitly
configured node_ip.
* Update config-sample.yaml
Bound the size of HTTP request bodies on the main API listener so large
messages cannot exhaust memory. Configurable via limit.max_api_request_body_size
(defaults to 10 MiB, 0 disables).
Co-authored-by: Claude Opus 4.8 (1M context) <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>
* 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.
* 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
* Added optional "Power of Two Random Choices" algorithm for the node selector sort_by feature. The current, default behavior of picking the lowest-valued node remains.
* Support for participant attributes
* move metadata setters to LocalParticipant
* address feedback
* forward error
* update go mod
* update attributes first
* 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
* Add control of playout delay
Add config to enable playout delay. The delay will be limited by
[min,max] in the config option and calculated by upstream & downstream
RTT.
* check protocol version to enable playout delay
* Move config to room, limit playout-delay update interval, solve comments
* Remove adaptive playout-delay
* Remove unused config
* API execution timeout is now configurable
In certain environments, it can take longer than the default 2s to
fully execute API requests. Making execution timeout a configurable option.
* do not expose api to YAML. internal for now.
Sapan Diwakar