web: add "how it works" write-up + link it from the chat header
Standalone blog-style page (matching chunkfive/monospace b&w UX) explaining the whole solution: amplitude-as-data, the no-copy-paste signaling relay, the multi-level modem, the Opus->G.711 codec fix and 440 Hz beat, the ACK/outbox reliability layer, and the Hamming(12,8)+Gray FEC. Linked from chat.html's header.
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<h1>zebra report</h1>
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<p class="sub">volume modem chatroom · e2e encrypted · webrtc carrier ·
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<a href="how-it-works.html" style="color:#555">how it works</a> ·
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<a href="/" style="color:#555">unturf</a></p>
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<div class="app">
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229
web/how-it-works.html
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web/how-it-works.html
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<!DOCTYPE html>
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<html lang="en">
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<head>
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<meta charset="UTF-8">
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<meta name="viewport" content="width=device-width,initial-scale=1">
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<title>zebra report — how it works</title>
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<style>
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@font-face {
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font-family: 'chunkfiveregular';
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src: url('fonts/chunkfive-regular-webfont.woff2') format('woff2'),
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url('fonts/chunkfive-regular-webfont.woff') format('woff');
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font-weight: normal; font-style: normal;
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}
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* { box-sizing: border-box; margin: 0; padding: 0; }
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body {
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font-family: monospace; background: #fff; color: #000;
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padding: 2rem; max-width: 820px; margin: 0 auto; line-height: 1.6;
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}
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h1 {
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font-family: 'chunkfiveregular', serif;
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font-size: 3rem; font-weight: normal;
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letter-spacing: 0.02em; line-height: 1; margin-bottom: 0.2rem;
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}
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.sub {
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font-size: 0.75rem; color: #555; margin-bottom: 2.5rem;
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letter-spacing: 0.05em; text-transform: uppercase;
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}
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.sub a { color: #555; }
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h2 {
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font-family: 'chunkfiveregular', serif;
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font-size: 1.3rem; font-weight: normal;
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border-bottom: 1px solid #000;
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padding-bottom: 0.3rem; margin: 2.4rem 0 0.9rem;
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}
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p { margin-bottom: 0.9rem; }
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ul, ol { margin: 0 0 0.9rem 1.4rem; }
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li { margin-bottom: 0.35rem; }
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code {
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background: #f0f0f0; padding: 0 0.2rem;
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font-size: 0.85em; border: 1px solid #ddd;
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}
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.lead { font-size: 1.05rem; }
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.note { font-size: 0.8rem; color: #555; }
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.cta {
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display: inline-block; margin: 0.4rem 0;
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background: #000; color: #fff; border: 1px solid #000;
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padding: 0.5rem 1.2rem; text-decoration: none; font-size: 0.9rem;
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}
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.cta:hover { background: #333; }
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.diagram {
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border: 1px solid #000; background: #fafafa;
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padding: 1rem; font-size: 0.72rem; line-height: 1.45;
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overflow-x: auto; white-space: pre; margin: 1rem 0;
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}
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hr { border: none; border-top: 1px solid #ddd; margin: 2.5rem 0; }
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.foot { font-size: 0.75rem; color: #777; margin-top: 2rem; }
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</style>
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</head>
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<body>
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<h1>zebra report</h1>
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<p class="sub">how it works · a volume-modem chatroom over webrtc ·
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<a href="./">open the chat</a> · <a href="/">unturf</a></p>
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<p class="lead">
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Zebra report is a two-person chatroom where your words never travel as network
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packets. Each message is encoded into the <em>loudness</em> of an audio tone,
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carried inside an encrypted WebRTC voice stream, and decoded back into text on
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the far side. A network observer sees an ordinary encrypted call. The chat
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lives in the amplitude.
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</p>
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<p>
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This page walks through the whole machine: how two browsers find each other
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with no copy-paste, how data rides on volume, why a speech codec nearly killed
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it, and the error correction that finally made it solid.
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</p>
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<a class="cta" href="./">▶ open the chat</a>
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<h2>1 · the idea: data in amplitude, not packets</h2>
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<p>
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The original zebra-report channel modulates a browser tab's PulseAudio volume:
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step the volume through values at a fixed clock rate and the other side reads
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those steps back as data, exactly like a dial-up modem. The payload lives in a
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side channel that the network stack never sees.
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</p>
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<p>
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The browser version generates a pure 440 Hz tone locally (no microphone)
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and swings its <em>output gain</em> per clock tick. That modulated tone becomes
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the outbound WebRTC audio track. Wireshark sees only encrypted SRTP; the words
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are in the amplitude envelope inside it.
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</p>
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<h2>2 · bootstrapping with no copy-paste</h2>
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<p>
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WebRTC needs the two peers to swap connection descriptions (SDP) before media
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flows. The usual way is to copy a blob of text from one device to the other and
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back. We replaced that with a tiny <strong>signaling relay</strong>:
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</p>
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<div class="diagram">A: rendezvous code ──▶ wss://cors-proxy.uncloseai.com/zebra-signal ◀── code :B
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(relay pairs the two, forwards encrypted SDP)
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both type the same code · no link · no QR · no paste</div>
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<p>
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Both peers type the same rendezvous code. The page derives an opaque room id
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and an encryption key from that code, connects to the relay, and the relay
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forwards the offer/answer between the two. The SDP is encrypted with the code
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<em>before</em> it leaves the page, so the relay only ever sees ciphertext and
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an opaque room id. It carries setup only; chat content never touches it.
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</p>
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<h2>3 · the carrier and the modem</h2>
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<p>
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At connection time each browser starts a 440 Hz tone and modulates its
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gain. Rather than a plain on/off (one bit per tick), the modem is
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<strong>multi-level</strong>: each clock tick holds the volume at one of N
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levels, so a symbol carries log₂(N) bits — 4 levels = 2 bits, 16
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levels = 4 bits per tick.
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</p>
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<p>
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Each byte is framed by a minimum-level <em>start</em> symbol and a
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maximum-level <em>stop</em> symbol. The receiver locks onto that high-to-low
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edge, then re-derives its own low and high amplitude reference from the start
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and stop of <em>every byte</em> — so the level mapping self-calibrates
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continuously and tolerates drift in the channel's gain.
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</p>
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<h2>4 · the codec trap</h2>
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<p>
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The first cross-machine attempts decoded nothing. The cause was the audio
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codec. WebRTC defaults to <strong>Opus</strong>, a perceptual speech codec that
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re-quantizes audio in 20 ms frames and effectively normalizes loudness.
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That is precisely the information our modem rides on, so Opus erased it.
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</p>
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<p>
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Two fixes restored a clean channel:
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</p>
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<ul>
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<li><strong>Pin the codec to G.711</strong> (<code>setCodecPreferences</code>).
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G.711 is memoryless companding — it preserves the amplitude envelope
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sample by sample, like the PulseAudio path, instead of smearing it.</li>
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<li><strong>Use a single 440 Hz tone.</strong> The carrier had used
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440 Hz left / 441 Hz right for a stereo mode, but G.711 is mono, so
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the two tones downmixed into a 1 Hz <em>beat</em> — a slow
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sinusoidal swell that drowned the levels. One frequency, no beat.</li>
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</ul>
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<h2>5 · making it reliable</h2>
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<p>
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Even on a clean channel, a wireless link is lossy. Three layers make the chat
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dependable:
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</p>
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<ul>
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<li><strong>Framing + CRC.</strong> Every frame carries a CRC32; a corrupted
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frame is detected and dropped rather than shown as garbage.</li>
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<li><strong>Delivery ACKs + outbox.</strong> A sent message is held in an
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outbox — not echoed into your log — until the far side
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acknowledges it (the ACK echoes the frame's CRC as a message id). Only then
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does it move into the log, marked delivered. The receiver de-dupes on CRC, so
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a lost ACK never double-shows a message.</li>
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<li><strong>Retransmit with backoff.</strong> An un-acknowledged message is
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resent, with exponential backoff so a bad stretch spaces out retries instead
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of hammering the channel.</li>
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</ul>
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<h2>6 · error correction (the eureka)</h2>
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<p>
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The last problem was the worst kind: messages mostly worked, then a frame would
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arrive one bit wrong, fail its CRC, get dropped, and the sender would retry
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forever. The channel was good enough to <em>almost</em> work.
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</p>
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<p>
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The fix is forward error correction. Each data byte is carried as a
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<strong>Hamming(12,8)</strong> codeword — 8 data bits plus 4 parity bits
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— inside one byte-frame. Any single flipped bit is located and corrected
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on the far side <em>before</em> the CRC check, so a marginal channel self-heals
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instead of dropping the frame.
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</p>
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<p>
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Hamming corrects single-bit errors, but a misread volume level can flip two
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bits at once. So the levels are <strong>Gray-coded</strong>: adjacent amplitude
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levels differ by exactly one bit. The most common error — reading a level
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as its neighbour — becomes a single-bit flip, which is exactly what
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Hamming repairs. Together they turned a channel that gave up into one that
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delivers.
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</p>
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<h2>7 · encryption & threat model</h2>
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<p>
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Chat content is end-to-end encrypted at the application layer, on top of
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WebRTC's own SRTP encryption. Two modes:
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</p>
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<ul>
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<li><strong>Passphrase</strong> — everyone types the same room phrase;
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a 256-bit AES-GCM key is derived with PBKDF2-SHA256 (600k iterations).</li>
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<li><strong>Pubkey</strong> — 1-to-1 ECDH (P-256) between two public
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keys.</li>
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</ul>
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<p class="note">
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Protects against a network observer reading chat content. Does not protect
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against another process on either endpoint recording the local audio, or an
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endpoint compromise — that local audio trust boundary is the surface this
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project documents.
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</p>
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<h2>8 · testing without two phones</h2>
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<p>
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The modem and protocol are validated by a Node test suite that runs the real
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shipped code — no browser and no second device required
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(<code>make test-web</code>). It covers the CRC and frame codec, the Hamming
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and Gray logic, a full multi-level modem roundtrip through a simulated noisy
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channel, and recovery of off-by-one symbol errors. Over three thousand
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assertions guard every change to the wire format.
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</p>
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<hr>
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<p>Type a rendezvous code on two devices and watch text cross on nothing but the
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loudness of a tone.</p>
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<a class="cta" href="./">▶ open the chat</a>
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<p class="foot">
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zebra report · volume modem chatroom ·
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<a href="/" style="color:#777">unturf</a>
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</p>
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</body>
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</html>
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