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.
