Prove video flows: an e2e suite with a real WebRTC viewer
e2e-edge.sh proves the camera signalling relay round-trips, and says so honestly - it posts a stub SDP with go2rtc absent and accepts a 503 that came back from the gateway. But "the offer round-trips" says nothing about whether anyone would see a picture, and it once read as more than it was: a stray go2rtc left over from an earlier run answered the stub with an HTTP 500 and the test still printed ok. Video had never been shown to flow. scripts/e2e-camera.sh runs the real backend, the real proxy and the real go2rtc against the H2C, and scripts/e2e-camera-peer.ts plays the browser's part with werift: createOffer, relay through the backend, setRemoteDescription, then ICE, DTLS and SRTP, counting RTP out the far end. Measured: 124-149 packets and ~119-147 KB of payload within a second or two. It sends no printer command; the camera is a pull and the MQTT subscription is passive. It refuses to start if a go2rtc is already running, and fails if one survives cleanup. werift is a devDependency for that one file; nothing under src/ imports it. Its 30 resolved packages were audited against LICENSES.md: 26 MIT, Apache-2.0, BSD-3-Clause, Unlicense, and one MPL-2.0 (mediabunny), which the policy allows while we do not edit its files. Smaller fixes in the same pass: - e2e-edge.sh now unsets the TURN variables. It asserts TURN is off, but sourcing dev-secrets.sh turns it on, and the failure message then printed the live Cloudflare username and credential. The new script prints only the camera id and a count, never the /stream body. - the last hardcoded `psql -h localhost` in e2e-edge.sh now uses DATABASE_URL, like the rest.
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/**
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* A WebRTC receiver that is not a browser, for scripts/e2e-camera.sh.
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*
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* What it proves that nothing else here does: that video actually arrives.
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* The signalling relay has always been testable with a canned SDP, and the
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* go2rtc-to-printer leg was confirmed by hand, but "the offer round-trips"
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* says nothing about whether a viewer would see a picture. This runs the
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* browser's half of the exchange for real - createOffer, relay through the
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* backend, setRemoteDescription, then ICE, DTLS and SRTP to go2rtc - and
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* counts the RTP packets that come out the other end.
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*
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* It also exists because the suite's earlier stub offer (`v=0 ... t=0 0`, no
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* media section, no ice-ufrag) could only ever be refused, and the test
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* counted that refusal as success. A real offer has to be accepted.
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*
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* Results are printed as `ok<TAB>label` / `FAILED<TAB>label`, the convention
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* run_probe in e2e-real-services.sh also reads. werift is a devDependency of
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* this repo for this file alone; nothing under src/ imports it.
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*
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* Environment: BASE, PRINTER_UUID, ACCESS_TOKEN. The token never reaches a
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* log line.
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*/
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import { RTCPeerConnection, RTCRtpCodecParameters } from "werift";
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const base = process.env.BASE!;
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const printer = process.env.PRINTER_UUID!;
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const token = process.env.ACCESS_TOKEN!;
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const MIN_PACKETS = Number(process.env.MIN_RTP_PACKETS ?? 50);
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const WAIT_MS = Number(process.env.MEDIA_WAIT_MS ?? 25_000);
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let failed = false;
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const say = (good: boolean, label: string, detail = "") => {
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if (!good) failed = true;
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console.log(`${good ? "ok" : "FAILED"}\t${label}${detail ? ` (${detail})` : ""}`);
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};
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// werift offers only VP8 by default, and a Bambu camera is H.264 (High, level
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// 4.1), so go2rtc refused the first version of this offer with "codecs not
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// matched: video:H264 => video:VP8" - correctly. Every browser offers H.264,
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// so offering it here is what a real viewer does, not a concession to the test.
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// The parameters are Chrome's: packetization-mode=1, constrained baseline.
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const pc = new RTCPeerConnection({
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iceServers: [],
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codecs: {
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video: [
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new RTCRtpCodecParameters({
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mimeType: "video/H264",
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clockRate: 90000,
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rtcpFeedback: [{ type: "nack" }, { type: "nack", parameter: "pli" }, { type: "goog-remb" }],
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parameters: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
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}),
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],
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},
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});
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const transceiver = pc.addTransceiver("video", { direction: "recvonly" });
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let packets = 0;
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let bytes = 0;
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transceiver.onTrack.subscribe((track) => {
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track.onReceiveRtp.subscribe((rtp) => {
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packets += 1;
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bytes += rtp.payload.length;
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});
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});
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const offer = await pc.createOffer();
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await pc.setLocalDescription(offer);
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const offerSdp = pc.localDescription!.sdp;
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say(/^m=video /m.test(offerSdp) && /^a=ice-ufrag:/m.test(offerSdp),
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"the offer has a video section and an ice-ufrag, unlike the stub it replaces");
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// A player that gets "stream not found" just asks again, and so does this: the
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// gateway advertises a camera as soon as the printer reports its URL, a moment
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// before go2rtc has been restarted with the stream in its config. Bounded, and
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// the attempt count is reported, so a window that grows shows up as a number
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// rather than as a flaky pass. Only that one reason is retried - a codec
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// mismatch or any other refusal is a real failure and must fail at once.
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const RETRY_MS = Number(process.env.OFFER_RETRY_MS ?? 20_000);
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const started = Date.now();
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let attempts = 0;
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let res: Response;
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let bodyText = "";
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do {
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attempts += 1;
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res = await fetch(`${base}/printers/${printer}/webrtc`, {
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method: "POST",
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headers: { authorization: `Bearer ${token}`, "content-type": "application/json" },
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body: JSON.stringify({ sdp: offerSdp }),
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});
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bodyText = await res.text();
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if (res.status !== 503 || !bodyText.includes("stream not found")) break;
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await Bun.sleep(1000);
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} while (Date.now() - started < RETRY_MS);
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say(res.status === 200, "the relayed offer is answered 200",
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`HTTP ${res.status} after ${attempts} attempt(s), ${Math.round((Date.now() - started) / 1000)}s`);
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if (res.status !== 200) {
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// The gateway's reason is the whole diagnosis, and it carries no secrets.
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console.log(`# ${bodyText.slice(0, 300).replace(/\s+/g, " ")}`);
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await pc.close();
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process.exit(1);
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}
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const answer = (JSON.parse(bodyText) as { sdp?: string }).sdp ?? "";
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say(/^m=video /m.test(answer) && /H264\/90000/i.test(answer),
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"the answer negotiates H264 video, which is what the printer sends");
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await pc.setRemoteDescription({ type: "answer", sdp: answer });
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const connected = await new Promise<boolean>((resolve) => {
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const timer = setTimeout(() => resolve(false), WAIT_MS);
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pc.connectionStateChange.subscribe((state) => {
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if (state === "connected") { clearTimeout(timer); resolve(true); }
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if (state === "failed") { clearTimeout(timer); resolve(false); }
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});
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if (pc.connectionState === "connected") { clearTimeout(timer); resolve(true); }
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});
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say(connected, "ICE and DTLS completed with go2rtc", `state=${pc.connectionState}`);
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if (connected) {
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const deadline = Date.now() + WAIT_MS;
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while (packets < MIN_PACKETS && Date.now() < deadline) await Bun.sleep(250);
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}
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say(packets >= MIN_PACKETS,
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`video RTP reached the viewer: at least ${MIN_PACKETS} packets`,
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`${packets} packets, ${bytes} payload bytes`);
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await pc.close();
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process.exit(failed ? 1 : 0);
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