# ZuckZapGo — Calls API: complete LLM build guide

> Paste this whole file into any LLM (Claude, GPT, Gemini, …) and ask it to build a
> WhatsApp **voice-call** integration against ZuckZapGo. It is self-contained: every
> endpoint, the exact audio wire-format, the enable sequence, full end-to-end flows, a
> drop-in browser client, a server-side (Node/Python/Go) client, the AI-voice-agent mode,
> and the common pitfalls are all here. Nothing else is required.
>
> Native two-way call audio powered by **meowcaller** (https://github.com/purpshell/meowcaller) — 🙏 thank you, Rajeh (@purpshell).

- **Format**: this document is the source of truth. Field names, status codes, and the PCM
  framing below are exact — copy them verbatim.
- **Two placeholders** you must fill in: `{BASE_URL}` (e.g. `https://your-host` or
  `http://localhost:8080`) and `{TOKEN}` (the per-instance user token).

---

## 0. The one-paragraph mental model

ZuckZapGo exposes a **native, pure-Go VoIP engine**. You drive calls with plain **REST**
(`/call/*`, JSON, token header) and you carry **audio** over a single **bidirectional
WebSocket** at `GET /call/{call_id}/stream`. That socket speaks **raw PCM**: signed
16-bit little-endian, **16 000 Hz, mono**, in **960-sample (1920-byte, 60 ms) frames**.
Inbound peer audio arrives as binary frames; you send the microphone back as binary frames.
There is **no WebRTC, no SDP, no ICE** to deal with — the engine already terminates the
WhatsApp media relay (SRTP) for you and hands you decoded PCM. Your only job is: enable the
engine, place/answer a call, open the WebSocket, and play/capture PCM.

```
                 control plane (REST, token header)
  your app  ───────────────────────────────────────────────►  ZuckZapGo  ──► WhatsApp
            POST /call/dial · /call/answer · /call/hangup · …    (VoIP engine)   relay
            ◄───────────────────────────────────────────────
                 media plane (WebSocket, ?token=)
  your app  ◄══════ s16le 16kHz mono PCM (peer audio) ════════  ZuckZapGo  ◄═ SRTP/16kHz
            ═══════ s16le 16kHz mono PCM (your mic)  ═══════►   /call/{id}/stream
```

**Golden rules**
1. Audio only flows **after the call is answered** (callee picks up an outbound call, or you
   answer an inbound call). Before that the relay is silent — that is normal, not a bug.
2. For app/browser-driven two-way audio, the instance **must** be in `call_inbound_mode:
   "manual"`. Other modes make the engine answer and handle audio server-side.
3. The audio WebSocket and server-side **recording are mutually exclusive per call** — they
   share the call's single inbound sink. Use one or the other.

---

## 1. Authentication

| Surface | How to authenticate |
|---|---|
| REST `/call/*` (and all standard endpoints) | HTTP header `token: {TOKEN}` |
| WebSocket `/call/{call_id}/stream` | Query param `?token={TOKEN}` (browsers cannot set WS headers). The `token:` header also works for non-browser clients. |
| Admin endpoints (not needed for calls) | HTTP header `Authorization: {ADMIN_TOKEN}` |

The token identifies the **instance** (one WhatsApp number). All `/call/*` calls operate on
that instance's engine.

---

## 2. Response envelope

Every JSON response is wrapped:

```jsonc
// success
{ "code": 200, "data": { /* payload */ }, "success": true }
// error
{ "code": 409, "error": "calls engine not enabled for this instance ...", "success": false }
```

Always read the payload from `.data` and check `.success` / HTTP status. Examples below show
the `data` payload.

---

## 3. Two call APIs — use the native engine one for audio

ZuckZapGo has **two** families under `/call/*`. Do not mix them up:

| Family | Endpoints | Purpose | Audio? |
|---|---|---|---|
| **Native VoIP engine** ✅ | `/call/config`, `/call/status`, `/call/dial`, `/call/answer`, `/call/hangup`, `/call/play`, `/call/record/start`, `/call/record/stop`, `/call/{id}/stream` | Real media: dial, answer, two-way PCM audio, play files, record, AI agents | **Yes** — this is the one you want |
| Legacy signaling (Spec 004) | `/call/reject`, `/call/accept`, `/call/preaccept`, `/call/terminate`, `/call/initiate`, `/call/reject/send` | Raw WhatsApp call-signaling control only | **No.** `/call/initiate` returns **501** (needs a WebRTC stack not in this build). Ignore these for audio. |

**To make calls with audio, only use the Native VoIP engine endpoints.** Outbound = `POST
/call/dial` (never `/call/initiate`).

---

## 4. Endpoint reference (Native VoIP engine)

All paths are relative to `{BASE_URL}`. All REST calls send `token: {TOKEN}`.

### 4.1 `GET /call/config` — read engine configuration

Response `data`:
```jsonc
{
  "callsEnabled": true,
  "callInboundMode": "manual",   // manual | bot | ivr | ai | webhook | reject
  "callRecord": false,
  "callSttUrl": "",              // AI modes only
  "callLlmUrl": "",
  "callTtsUrl": "",
  "callSystemPrompt": "",
  "callGreeting": ""
}
```
> `callProviderToken` is write-only and never returned.

### 4.2 `PUT /call/config` — update engine configuration (applies on next reconnect)

Body:
```jsonc
{
  "callsEnabled": true,
  "callInboundMode": "manual",      // empty defaults to "webhook"; invalid value → 400
  "callRecord": false,
  "callSttUrl": null,               // AI modes (ai/ivr/bot); see §9
  "callLlmUrl": null,
  "callTtsUrl": null,
  "callProviderToken": null,        // bearer token for the STT/LLM/TTS provider(s)
  "callSystemPrompt": null,
  "callGreeting": null
}
```
Response `data`: `{ "ok": true }`. **The new config takes effect on the instance's next
reconnect** — see §5.

### 4.3 `GET /call/status` — list live calls

Response `data`:
```jsonc
{
  "calls": [
    {
      "callId": "A1B2C3...",        // engine's own id — use this everywhere
      "peer": "5521999999999@s.whatsapp.net",
      "state": "active",            // idle|calling|ringing|connecting|active|ended|unknown
      "direction": "inbound",       // inbound | outbound
      "video": false,
      "recording": false,
      "startedAt": "2026-06-28T12:00:00Z"
    }
  ]
}
```
Poll this (~2.5 s) to detect **inbound ringing calls** and remote hangups — the media socket
alone does not signal a remote hangup. **Detect "call ended" by the call's _disappearance_ from
the `calls[]` array, not by a `state:"ended"` value**: when a call ends the engine stops
tracking it, so it vanishes from the list (you will rarely, if ever, observe `state:"ended"`
here). Once a call you were in is gone from `calls[]`, tear your UI/socket down.

### 4.4 `POST /call/dial` — place an outbound call

Body: `{ "phone": "5521999999999", "video": false }`
- `phone` = **E.164 digits only, no `+`** (country code + number).
- `video` is **accepted but currently ignored** — outbound is voice-only in this build. Do not
  build a video-dial UI against it (it is a no-op, not an error).
- Response `data`: `{ "callId": "A1B2C3..." }`.
- **409** if the engine is not enabled (enable + reconnect first, §5).
- **502** if the peer is unreachable.

After dialing, open the media WebSocket (§6) immediately; audio starts when the callee answers.

### 4.5 `POST /call/answer` — answer a ringing inbound call (manual mode)

Body: `{ "callId": "A1B2C3..." }` → `data: { "ok": true }`. **404** if the call is not found.
Then open the media WebSocket (§6).

### 4.6 `POST /call/hangup` — end a live call

Body: `{ "callId": "A1B2C3..." }` → `data: { "ok": true }`. **404** if not found. Also close
the WebSocket. Use this to **decline** an inbound call too.

### 4.7 `POST /call/play` — play an audio file/clip into the call

Body: `{ "callId": "...", "audioUrl": "https://…/clip.mp3" }`
or `{ "callId": "...", "audioBase64": "<base64 of WAV/MP3/Ogg-Opus>" }`.
- Format auto-detected (WAV / MP3 / Ogg-Opus) and resampled to 16 kHz mono.
- Plays to the peer. → `data: { "ok": true }`. Works alongside or instead of the mic stream.

### 4.8 `POST /call/record/start` — start server-side recording

Body: `{ "callId": "..." }` → `data: { "ok": true }`. Records the **peer's** audio to a WAV.
Mutually exclusive with the media WebSocket on the same call. **`start` succeeds regardless of
storage config** — a missing/misconfigured S3 does not fail here; it surfaces later as an empty
`mediaKey` on `stop` (see §4.9).

### 4.9 `POST /call/record/stop` — stop + upload recording

Body: `{ "callId": "..." }` → `data: { "ok": true, "mediaKey": "<storage key>" }`. The WAV is
uploaded to object storage (S3) and `mediaKey` is its key. **Treat an empty `mediaKey` (`""`)
as failure** — `ok` is still `true` even when the upload had no storage configured or errored,
so check `mediaKey` is non-empty, not just `ok`.

### 4.10 `GET /call/{call_id}/stream` — bidirectional PCM media WebSocket

The audio plane. Full protocol in §6. Auth via `?token={TOKEN}`.

---

## 5. Enabling the engine (do this once per instance)

The engine attaches when the instance connects. To turn it on for app-driven audio:

```bash
# 1) Enable native calls in MANUAL mode (required for two-way app/browser audio)
curl -X PUT "{BASE_URL}/call/config" \
  -H "token: {TOKEN}" -H "Content-Type: application/json" \
  -d '{"callsEnabled":true,"callInboundMode":"manual","callRecord":false}'

# 2) Reconnect so the engine attaches — subscribe to "Call" so inbound offers ring.
#    (Read current events first to preserve existing subscriptions.)
curl -X POST "{BASE_URL}/session/disconnect" -H "token: {TOKEN}" -d '{}'
sleep 1
curl -X POST "{BASE_URL}/session/connect" \
  -H "token: {TOKEN}" -H "Content-Type: application/json" \
  -d '{"Subscribe":["Message","Call"],"Immediate":true}'
```

After reconnect, `GET /call/status` works and `POST /call/dial` no longer returns 409.
**Why reconnect**: `callsEnabled` / `call_inbound_mode` are read when the engine attaches at
connect time. Without a reconnect a running session keeps the old (disabled) engine.

Optional (push instead of polling): point the instance webhook at your server and subscribe to
the `Call` event. Inbound calls then emit `v1.call.*` webhook events — the inbound **ring** is
**`v1.call.offer`** (others: `v1.call.offer_notice`, `v1.call.accept`, `v1.call.preaccept`,
`v1.call.transport`, `v1.call.terminate`, `v1.call.reject`, `v1.call.relay_latency`,
`v1.call.unknown`). These come from the **standard `Call` event subscription** and are
independent of the legacy `/call/initiate` signaling endpoints (§3) — you do **not** need that
legacy path to receive them. Match the webhook's call id to `/call/status`, then `POST
/call/answer`. Full payload shapes: see `docs/webhook-events.md`. If you don't want webhooks,
polling `/call/status` (§4.3) is a complete alternative.

---

## 6. The media WebSocket protocol (exact)

```
GET {BASE_URL_AS_WS}/call/{call_id}/stream?token={TOKEN}
```
- `{BASE_URL_AS_WS}` = your base URL with `http`→`ws` / `https`→`wss`.
- `{call_id}` = the `callId` returned by `/call/dial` or seen in `/call/status` (URL-encode it).
- The connection is a standard WebSocket. Use **binary** frames only; text frames are ignored.

**Audio format on the wire (both directions): raw PCM, no header, no container.**

| Property | Value |
|---|---|
| Sample format | signed 16-bit **little-endian** (`s16le` / `int16`) |
| Sample rate | **16 000 Hz** |
| Channels | **1 (mono)** |
| Frame size | **960 samples = 1920 bytes = 60 ms** |

- **Server → client** (binary): decoded **peer** audio. Each message is s16le PCM. Treat the
  payload as a stream of `int16` little-endian samples; lengths are multiples of 2 bytes.
  Play them out (use a small jitter buffer — see §7).
- **Client → server** (binary): your **microphone** audio, s16le 16 kHz mono. **Send exactly
  960-sample (1920-byte) frames.** The server splits incoming bytes into 960-sample frames and
  **zero-pads** any trailing partial frame to 960 samples (injecting a little silence into the
  call) — so always frame to exactly 1920 bytes before sending to avoid that.

**PCM sample conversions** (the only math you need):
```js
// int16 little-endian  ->  float32 in [-1, 1]
function s16leToFloat(arrayBuffer) {
  const v = new DataView(arrayBuffer), n = (arrayBuffer.byteLength >> 1);
  const out = new Float32Array(n);
  for (let i = 0; i < n; i++) out[i] = v.getInt16(i * 2, true) / 0x8000;
  return out;
}
// float32 in [-1, 1]  ->  int16 little-endian
function floatToS16LE(frame) {
  const buf = new ArrayBuffer(frame.length * 2), v = new DataView(buf);
  for (let i = 0; i < frame.length; i++) {
    const s = Math.max(-1, Math.min(1, frame[i] || 0));
    v.setInt16(i * 2, s < 0 ? s * 0x8000 : s * 0x7fff, true);
  }
  return buf;
}
```

The socket closes when the call ends or you call `/call/hangup`. A remote hangup does **not**
always close the socket promptly — also watch `/call/status` (poll) to tear your UI down.

---

## 7. Drop-in browser client (Web Audio / AudioWorklet)

This is the validated browser path: an `AudioContext` at 16 kHz, a **player** AudioWorklet
(jitter buffer, silence on underrun) for inbound audio, and a **recorder** AudioWorklet that
posts mic frames you batch into 1920-byte frames and send. A linear resampler is used only if
the browser refuses a 16 kHz context.

> A complete, production widget already ships in this repo at
> `/dashboard` → `static/dashboard/js/calls.js`. Read it as the reference. The essentials:

```js
const SAMPLE_RATE = 16000, FRAME_SAMPLES = 960; // 60 ms @ 16 kHz

// --- inlined AudioWorklet processors (loaded as Blob modules) ---
const PLAYER_WORKLET = `
class PcmPlayer extends AudioWorkletProcessor {
  constructor(){ super(); this.q=[]; this.r=null; this.p=0; this.b=0; this.play=false;
    this.pre=Math.floor(sampleRate*0.18); this.max=sampleRate*1;   // ~180ms prebuffer, 1s cap
    this.port.onmessage=(e)=>{ const d=e.data;
      if(d==='flush'){ this.q=[]; this.r=null; this.p=0; this.b=0; this.play=false; return; }
      if(this.b>this.max){ this.q=[]; this.r=null; this.p=0; this.b=0; }
      this.q.push(d); this.b+=d.length; }; }
  pull(){ if(!this.r||this.p>=this.r.length){ this.r=this.q.shift()||null; this.p=0; if(!this.r) return 0; } this.b--; return this.r[this.p++]; }
  process(_i,o){ const out=o[0]&&o[0][0]; if(!out) return true;
    if(!this.play){ if(this.b>=this.pre) this.play=true; else { out.fill(0); return true; } }
    for(let i=0;i<out.length;i++){ if(this.b<=0){ this.play=false; out[i]=0; } else { out[i]=this.pull(); } } return true; } }
registerProcessor('pcm-player', PcmPlayer);`;

const RECORDER_WORKLET = `
class PcmRec extends AudioWorkletProcessor {
  constructor(){ super(); this.muted=false;
    this.port.onmessage=(e)=>{ if(e.data&&typeof e.data.muted==='boolean') this.muted=e.data.muted; }; }
  process(inp){ const input=inp[0]&&inp[0][0];
    if(input&&input.length){ const c=new Float32Array(input.length); if(!this.muted) c.set(input);
      this.port.postMessage(c,[c.buffer]); } return true; } }
registerProcessor('pcm-recorder', PcmRec);`;

const blobModuleURL = (code) => URL.createObjectURL(new Blob([code], { type: "application/javascript" }));

async function startCallAudio(baseUrl, callId, token) {
  const wsUrl = baseUrl.replace(/^http/i, "ws") + "/call/" + encodeURIComponent(callId) +
                "/stream?token=" + encodeURIComponent(token);
  const ctx = new (window.AudioContext || window.webkitAudioContext)({ sampleRate: SAMPLE_RATE });
  await ctx.resume();
  await ctx.audioWorklet.addModule(blobModuleURL(PLAYER_WORKLET));
  await ctx.audioWorklet.addModule(blobModuleURL(RECORDER_WORKLET));

  const player = new AudioWorkletNode(ctx, "pcm-player", { numberOfInputs:0, numberOfOutputs:1, outputChannelCount:[1] });
  player.connect(ctx.destination);

  const localStream = await navigator.mediaDevices.getUserMedia({
    audio: { channelCount:1, echoCancellation:true, noiseSuppression:true } });
  const recorder = new AudioWorkletNode(ctx, "pcm-recorder", { numberOfInputs:1, numberOfOutputs:0 });
  ctx.createMediaStreamSource(localStream).connect(recorder);

  const ws = new WebSocket(wsUrl); ws.binaryType = "arraybuffer";
  let acc = new Float32Array(0);

  // mic -> batch to 960-sample frames -> send s16le
  recorder.port.onmessage = (e) => {
    if (ws.readyState !== WebSocket.OPEN) return;
    const merged = new Float32Array(acc.length + e.data.length);
    merged.set(acc); merged.set(e.data, acc.length);
    let off = 0;
    while (merged.length - off >= FRAME_SAMPLES) {
      ws.send(floatToS16LE(merged.subarray(off, off + FRAME_SAMPLES)));
      off += FRAME_SAMPLES;
    }
    acc = merged.slice(off);
  };
  // peer audio -> player jitter buffer
  ws.onmessage = (ev) => {
    if (typeof ev.data === "string") return;
    const pcm = s16leToFloat(ev.data);
    player.port.postMessage(pcm, [pcm.buffer]);
  };

  return {
    setMuted: (m) => recorder.port.postMessage({ muted: m }),
    close: () => { try { ws.close(); } catch(e){} localStream.getTracks().forEach(t=>t.stop()); ctx.close().catch(()=>{}); },
  };
}
```

**Outbound from the browser**
```js
const r = await fetch(`${BASE_URL}/call/dial`, {
  method:"POST", headers:{ "Content-Type":"application/json", token: TOKEN },
  body: JSON.stringify({ phone: "5521999999999" }) });
const callId = (await r.json()).data.callId;
const session = await startCallAudio(BASE_URL, callId, TOKEN);   // callee answers -> audio
// ... session.setMuted(true) to mute ...
// hang up:
await fetch(`${BASE_URL}/call/hangup`, { method:"POST",
  headers:{ "Content-Type":"application/json", token: TOKEN }, body: JSON.stringify({ callId }) });
session.close();
```

**Inbound in the browser**: poll `GET /call/status`; when a call appears with
`direction:"inbound"` and `state` in `ringing|connecting|calling`, show a ring UI; on accept
`POST /call/answer {callId}` then `startCallAudio(BASE_URL, callId, TOKEN)`. Note: device
labels are blank until the page holds mic permission — call `getUserMedia` once to populate
`enumerateDevices()` labels if you build a device picker.

---

## 8. Server-side client (Node / Python / Go) — softphones, bots, AI

You don't need a browser. Any WebSocket client can bridge the PCM both ways — feed it from a
TTS engine, an AI voice agent, a SIP gateway, a file, etc. Read **1920-byte** binary frames
in; send **1920-byte** binary frames out.

**Node.js (`ws`)**
```js
import WebSocket from "ws";
const ws = new WebSocket(
  `${BASE_URL.replace(/^http/, "ws")}/call/${encodeURIComponent(callId)}/stream?token=${TOKEN}`
);
ws.binaryType = "nodebuffer";
ws.on("message", (buf) => {
  // buf = s16le 16kHz mono peer audio. Int16Array view:
  const pcm = new Int16Array(buf.buffer, buf.byteOffset, buf.byteLength >> 1);
  // -> write to speaker / STT / file ...
});
// send your audio: chunk your s16le 16kHz mono stream into 1920-byte Buffers
function sendFrame(int16Frame /* length 960 */) {
  ws.send(Buffer.from(int16Frame.buffer));   // 1920 bytes
}
```

**Python (`websockets`)**
```python
import asyncio, websockets, numpy as np
async def run(base_ws, call_id, token):
    url = f"{base_ws}/call/{call_id}/stream?token={token}"  # base_ws uses ws:// or wss://
    async with websockets.connect(url, max_size=None) as ws:
        async for msg in ws:                 # bytes = s16le 16kHz mono peer audio
            pcm = np.frombuffer(msg, dtype="<i2")   # int16 little-endian
            # ... play / STT ...
            # to talk back, send exact 1920-byte frames:
            # await ws.send(frame_int16_960.astype("<i2").tobytes())
```

This is the path for building **AI voice agents in your own stack**: peer audio → your STT →
your LLM → your TTS → 1920-byte frames back. (Or let ZuckZapGo run the loop for you — §9.)

---

## 9. Built-in AI / scripted inbound modes (no WebSocket needed)

Instead of `manual`, the engine can answer inbound calls and run audio server-side. Set these
via `PUT /call/config` (`callInboundMode` + the URLs/prompt). Audio is handled inside
ZuckZapGo; you observe via webhooks.

| `callInboundMode` | Behavior | Needs |
|---|---|---|
| `manual` | Your app drives answer/play/record and the PCM WebSocket. **Use for two-way browser/app audio.** | — |
| `webhook` | Engine auto-answers; lifecycle + (optional) recording surfaced via webhooks. | — |
| `bot` | Plays a scripted TTS greeting, then hangs up. | `callTtsUrl`, `callGreeting`, `callProviderToken` |
| `ivr` | Greeting + one listen/transcribe turn, surfaced via webhook. | `callSttUrl`, `callTtsUrl`, `callProviderToken` |
| `ai` | Full **STT → LLM → TTS** conversation loop on the server. | `callSttUrl`, `callLlmUrl`, `callTtsUrl`, `callSystemPrompt`, `callProviderToken` |
| `reject` | Engine off (call rejection handled by the legacy path). | — |

Example — turn the number into a server-side AI voice agent:
```bash
curl -X PUT "{BASE_URL}/call/config" -H "token: {TOKEN}" -H "Content-Type: application/json" -d '{
  "callsEnabled": true,
  "callInboundMode": "ai",
  "callSttUrl":  "https://your-stt-endpoint",
  "callLlmUrl":  "https://your-llm-endpoint",
  "callTtsUrl":  "https://your-tts-endpoint",
  "callProviderToken": "sk-...",
  "callSystemPrompt": "You are a helpful WhatsApp voice assistant. Be concise.",
  "callGreeting": "Hi! How can I help you today?"
}'
# then reconnect (see §5)
```
If an AI mode is missing required providers, the engine falls back to greeting + hangup.

### 9.1 Provider HTTP contracts (BYO STT / LLM / TTS) — you must implement these

In `bot`/`ivr`/`ai` modes ZuckZapGo calls **your** HTTP endpoints. They are plain JSON/audio
requests (no AI-SDK assumed). Implement them exactly like this — these are the real wire
contracts the engine speaks:

**Common auth**: when `callProviderToken` is set, every request carries
`Authorization: Bearer {callProviderToken}`.

**STT — `POST {callSttUrl}`** (speech → text)
- Request: `Content-Type: audio/wav`, body = a **WAV file, 16 kHz mono** (the captured utterance).
- Response: JSON `{ "text": "<transcript>" }`.

**LLM — `POST {callLlmUrl}`** (chat turn → reply)
- Request: `Content-Type: application/json`, body:
  ```jsonc
  { "system": "<callSystemPrompt>",
    "messages": [ { "role": "user", "content": "..." }, { "role": "assistant", "content": "..." } ] }
  ```
  (full conversation history; roles are `user` / `assistant`).
- Response: JSON `{ "reply": "<assistant text>" }`.

**TTS — `POST {callTtsUrl}`** (text → speech)
- Request: `Content-Type: application/json`, `Accept: audio/wav`, body `{ "text": "<to speak>" }`.
- Response: **either** a WAV file (`RIFF`/`WAVE` — any sample rate, resampled for you) **or**
  raw **s16le 16 kHz mono PCM** (anything that isn't a RIFF header is treated as raw PCM).

A non-2xx from any provider aborts that step. If you'd rather own the whole loop, use `manual`
mode + the §8 WebSocket client instead of these endpoints.

---

## 10. End-to-end recipes

**A) Outbound call with two-way audio**
1. (once) `PUT /call/config {callsEnabled:true, callInboundMode:"manual"}` → reconnect (§5).
2. `POST /call/dial {phone}` → `callId`.
3. Open `wss://…/call/{callId}/stream?token=…`; start the AudioWorklet/player+recorder.
4. Callee answers → audio both ways. `setMuted(true)` to mute.
5. `POST /call/hangup {callId}` + close the socket.

**B) Inbound call with two-way audio**
1. (once) manual mode + reconnect, subscribe `Call`.
2. Detect ring: the **`v1.call.offer`** webhook (Call subscription) **or** poll
   `GET /call/status` for `direction:"inbound"`, `state in ringing|connecting|calling`.
3. `POST /call/answer {callId}` → open the socket → audio.
4. Decline instead = `POST /call/hangup {callId}`.

**C) Play a clip / IVR-style prompt without a mic**
1. Dial or answer as above (manual mode).
2. `POST /call/play {callId, audioUrl|audioBase64}` (WAV/MP3/Ogg-Opus). No WebSocket required
   if you only need outbound audio.

**D) Record a call**
1. Ensure S3 is configured for the instance.
2. `POST /call/record/start {callId}` … `POST /call/record/stop {callId}` → `mediaKey`.
   (Do **not** also open the PCM WebSocket on the same call — mutually exclusive.)

**E) Server-side AI agent (ZuckZapGo-managed)**: §9 `ai` mode — no socket, no app audio code.

**F) Server-side AI agent (your stack)**: manual mode + the §8 server WebSocket client.

---

## 11. Pitfalls & troubleshooting (read this)

- **`409 calls engine not enabled`** → you enabled config but didn't reconnect. Do §5.
- **Dial works but no audio** → (a) the callee hasn't answered yet (audio starts on answer);
  (b) you're not in `manual` mode; (c) browser `AudioContext` is suspended — call `ctx.resume()`
  after a user gesture; (d) the player worklet isn't loaded / WS isn't `binaryType:"arraybuffer"`.
- **Choppy / robotic audio** → you're not framing outbound to exactly **1920 bytes**, or your
  player has no jitter buffer. Batch to 960 samples; prebuffer ~180 ms.
- **Inbound rings but `/answer` 404s** → use the `callId` exactly as returned by `/call/status`
  (it is the engine's own id; do not transform case).
- **Remote hangup leaves UI stuck** → also poll `/call/status`; the socket may not close
  promptly. The ended call **disappears** from `calls[]` (you won't see `state:"ended"`).
- **Recording produced no file / empty `mediaKey`** → `/call/record/start` always returns `ok`,
  but the WAV upload needs S3 configured; a missing/failed upload yields `mediaKey:""` on
  `/call/record/stop` (with `ok:true`). Treat empty `mediaKey` as failure. Also, the call must
  not have an open PCM WebSocket (recording and `/stream` share the single inbound sink).
- **WS connects then closes immediately** → bad/missing `?token=`, or the `call_id` isn't a live
  call (check `/call/status`).
- **Server-side debugging**: set env `VOIP_LOG_LEVEL=debug` on ZuckZapGo and watch logs tagged
  `subsystem:calls` — `offer sent`, `relay silent after allocate`, `starting media`,
  `first RTP decoded from relay`, `selected audio codec`, `failed to unprotect`. `offer sent`
  then silence = callee never answered (not a bug).

---

## 12. Quick reference

```
AUTH        REST: header  token: {TOKEN}        WS: query  ?token={TOKEN}
ENVELOPE    { code, data, success }  |  errors: { code, error, success:false }

REST (Native VoIP engine)
  GET  /call/config                      -> engine config
  PUT  /call/config                      {callsEnabled,callInboundMode,callRecord,call*Url,callProviderToken,callSystemPrompt,callGreeting} (reconnect to apply)
  GET  /call/status                      -> { calls:[{callId,peer,state,direction,video,recording,startedAt}] }
  POST /call/dial                        {phone (E.164 no +), video(ignored, voice-only)} -> {callId}  (409 if disabled, 502 if unreachable)
  POST /call/answer                      {callId} -> {ok}             (404 if not found)
  POST /call/hangup                      {callId} -> {ok}             (also: decline)
  POST /call/play                        {callId, audioUrl|audioBase64} -> {ok}
  POST /call/record/start                {callId} -> {ok}            (excl. with /stream; succeeds even w/o S3)
  POST /call/record/stop                 {callId} -> {ok, mediaKey}  (empty mediaKey = upload failed/no S3)

MEDIA (WebSocket)
  GET  /call/{call_id}/stream?token=...  bidirectional binary PCM
       format: s16le, 16000 Hz, mono     frame: 960 samples = 1920 bytes = 60 ms
       server->client: peer audio        client->server: your mic (send exact 1920-byte frames)

INBOUND     v1.call.offer webhook (Call subscription)  OR  poll /call/status
MODES       manual(app audio) | webhook | bot | ivr | ai | reject
STATES      idle | calling | ringing | connecting | active | (ended = call drops from /call/status)
DIRECTION   inbound | outbound
```

---

## 13. Copy-paste task prompt for your LLM

> You are integrating WhatsApp **voice calls** into <my app / stack> using the ZuckZapGo
> Calls API documented above. Build <a browser softphone | a server-side voice bot | an AI
> voice agent>. Requirements:
> 1. Enable the engine once (`PUT /call/config` `manual` mode) and reconnect.
> 2. Outbound via `POST /call/dial`; inbound via `/call/status` polling or the `v1.call.offer`
>    webhook + `POST /call/answer`.
> 3. Carry audio over `GET /call/{call_id}/stream?token=…` as **s16le 16 kHz mono PCM**, sending
>    the mic in **exact 1920-byte (960-sample, 60 ms) frames** and playing inbound frames with a
>    ~180 ms jitter buffer.
> 4. Mute = send **zero-filled (silence)** frames, keeping the socket open (the reference does
>    this — it does not stop the send loop); hang up = `POST /call/hangup` + close the socket;
>    detect remote hangups by the call **disappearing** from `/call/status`.
> 5. Use the response envelope `{code,data,success}` and the `token` header (`?token=` for the WS).
> Produce complete, runnable code for <language/framework>, with reconnect/cleanup handling.

---

_Powered by meowcaller (https://github.com/purpshell/meowcaller). For the in-repo backend
details and the production-validated rationale, see `docs/call-audio-implementation-guide.md`._