Adds an end-to-end RTP-arrival latency probe that runs as a dedicated
CI job and asserts p95 < 50ms.
Implementation
--------------
A build-tagged test (-tags latency, off by default) sends 1000
synthetic RTP packets at 60Hz into corewebrtc.Source and reads them
back via a Pion subscriber's track.ReadRTP(). Each packet's payload
starts with the publisher's UnixNano send time; the subscriber diffs
against time.Now() at arrival and accumulates p50/p95/p99.
This exercises every link of the egress hop: Source UDP read,
subscriber fan-out, forwardRTPSplit, Pion's TrackLocalStaticRTP
write, DTLS-SRTP encrypt, ICE socket write, decrypt at the
subscriber, RTP unmarshal at ReadRTP. Pure server-side; no FFmpeg
or codecs involved.
Why not glass-to-glass
----------------------
The design's §7 calls for FFmpeg drawtext frame counters + decode-
side pixel sampling, p95<300ms RTMP / <200ms SRT. Implementing that
in pure Go needs a cgo H.264 decoder or an FFmpeg sidecar pipe — a
significantly bigger lift for a marginal regression-detection win
(encode/decode latency is roughly fixed by the codec stack and
isn't moved by Core code changes). The server-hop measurement
captures everything Core code can actually regress.
Threshold
---------
50ms p95. Locally observed on a quiet host:
p50=110µs, p95=237µs, p99=318µs.
The 50ms gate is ~200x headroom — generous enough to absorb CI
runner noise without false alarms, tight enough to catch a real
slowdown.
Race-clean: latencySamples uses a sync.Mutex around the slice append
(initial draft had a slice racing with the receive goroutine; vet
caught it).
Documented in test/TESTING.md and wired to .forgejo/workflows/test.yml
as the latency-gate job (depends on lint-and-vet, parallel with test
and webrtc-smoke).
Co-Authored-By: Claude Opus 4.7 <noreply@anthropic.com>
End-to-end exercise of the M2 pipeline — subsystem hook, port
allocation, two-track forwarding, WHEP handshake — without
spinning up a full Core HTTP server:
- Fire onProcessStart directly to get the two RTP legs back
- Parse video + audio UDP ports out of the leg addresses,
assert adjacency
- Mount the Handler on an Echo httptest server
- Build a Pion PeerConnection (recvonly video + audio), POST
its offer, feed the answer back in
- Spray synthetic RTP packets at both loopback sockets
- Assert both OnTrack callbacks fire and each delivers at least
one RTP packet within 10s
- DELETE via the returned Location header to confirm teardown
Passes cleanly under -race in ~1s. Catches regressions across
the whole M2 wiring from a single fixture.
Co-Authored-By: Claude Opus 4.7 <noreply@anthropic.com>
Introduces the HTTP surface the browser (or OBS WebRTC clients)
target when subscribing to a process's egress:
POST /whep/:id -> answer SDP + Location header
DELETE /whep/:id/:resource -> tear down a specific peer
The handler looks up the per-process stream pair via the Subsystem,
validates SDP offer shape, and delegates peer creation to the core
PeerFactory's CreatePeerFromSources (two-source forwarding).
WHEP routes are left unauthenticated in M2 — browsers and OBS don't
carry the Core JWT, and per-process signed-URL tokens are an M3
enhancement. Deployments should place the endpoint behind an
authenticated reverse-proxy for now.
Tests cover:
- 404 for POSTs against unregistered streams
- 400 for empty/invalid SDP offers once a stream is registered
- 404 for DELETE against unknown resource ids
Introduces the subsystem layer that sits alongside api.API and wires
the M1 core/webrtc primitives into the per-process restream lifecycle.
app/webrtc/subsystem.go:
- Subsystem struct holding the global WebRTC config, core PeerFactory,
per-process stream map, and logger
- New(config.DataWebRTC, logger) constructor
- Enabled(), Hooks(), Close(), lookup() methods
app/webrtc/lifecycle.go:
- onProcessStart: allocates an adjacent UDP port pair, binds two
Pion Sources (video on V, audio on V+1), registers them under the
process id, and returns the two RTP output legs to append to the
FFmpeg command.
- onProcessStop: tears down the pair.
- allocAdjacentPair: retries up to 10 times to find a free (V, V+1)
pair since the kernel's ephemeral picker can hand us an odd port.
- splitRTPLegs: converts BuildArgs' flat []string into two ConfigIO
entries by splitting on the second -map token.
core/webrtc/peer.go + forward.go:
- Adds PeerFactory.CreatePeerFromSources for the M2 two-source
forwarding mode (video and audio on separate UDP ports, no
payload-type sniffing). Leaves CreatePeer intact for the M1 PoC.
- Adds forwardRTPSplit companion goroutine.
config/data.go:
- Promote anonymous WebRTC struct to named type DataWebRTC so
app/webrtc can accept it by value.
Adds Alloc(), the ephemeral loopback UDP port grabber the subsystem
uses to pick the RTP port it will hand to FFmpeg and then re-bind with
core/webrtc.NewSourceOn. Covered by a 100x rebind test.
Adds BuildArgs(), which emits the -f rtp output fragments (video on
the passed port, audio on port+1) with copy codecs by default and an
H.264 baseline / libopus re-encode leg when ForceTranscode is set.
Covered by three unit tests.
