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@mrcall/directvoice

v1.0.7

Published

Browser client for MrCall DirectVoice — WebSocket voice calls with an AI assistant. Opus via WebCodecs, zero dependencies.

Readme

MrCallDirectVoice JS SDK

Browser SDK for embedding MrCall DirectVoice AI voice calls on any website. Zero dependencies. Single file. Supports Opus and PCM16.

Bundles

| File | Size | Description | |------|------|-------------| | MrCallDirectVoice.bundle.js | ~26 KB | SDK only, debug | | MrCallDirectVoice.bundle.min.js | ~17 KB | SDK only, minified + obfuscated | | MrCallDirectVoiceWidget.bundle.js | ~45 KB | Widget + SDK, debug | | MrCallDirectVoiceWidget.bundle.min.js | ~33 KB | Widget + SDK, minified + obfuscated |

Audio Encoding

| Encoding | Bandwidth | Browser requirement | Description | |----------|-----------|---------------------|-------------| | opus (default) | ~24 kbps | WebCodecs API (Chrome 94+, Edge 94+, Safari 16.4+) | Compressed audio. ~10x smaller than PCM. Uses browser-native WebCodecs encoder/decoder. | | pcm16 | ~384 kbps | Any modern browser | Raw 16-bit PCM. Universal fallback. |

Opus is enabled by default. If WebCodecs is not available, the SDK automatically falls back to PCM16 — no code changes needed.


Widget — One-line Embed

The easiest way to add MrCall DirectVoice to any website. A single <script> tag — no JavaScript needed.

Minimal embed

<script src="MrCallDirectVoiceWidget.bundle.min.js"
  data-base-url="https://your-server.com"
  data-auth-user="API_USER"
  data-auth-password="API_PASS"
  data-business-id="YOUR_BUSINESS_ID"
></script>

Full embed with customization

<script src="MrCallDirectVoiceWidget.bundle.min.js"
  data-base-url="https://your-server.com"
  data-auth-user="API_USER"
  data-auth-password="API_PASS"
  data-business-id="YOUR_BUSINESS_ID"
  data-username="end-user-id"
  data-email="[email protected]"
  data-display-name="Mario Rossi"
  data-title="Talk to our AI"
  data-bot-name="MrCall Assistant"
  data-color="#22c55e"
  data-popup-message="Need help? Click to talk!"
  data-show-popup="true"
  data-show-popup-time="3"
  data-position="bottom-right"
></script>

Widget data-* attributes

| Attribute | Default | Description | |-----------|---------|-------------| | data-base-url | — | Server URL (required) | | data-ws-url | — | WebSocket path (Option A: backend-initialized) | | data-auth-user | — | API username (Option B: full call) | | data-auth-password | — | API password | | data-business-id | — | CRM business ID | | data-username | — | End-user identifier | | data-email | — | End-user email | | data-display-name | — | End-user display name | | data-encoding | "opus" | "opus" or "pcm16" | | data-jitter-buffer-ms | "80" | Audio buffered before playback starts. 0 restores instant playback. | | data-language | — | ISO language tag (e.g. it-IT, en-US). Overrides business default. | | data-mode | "floating" | "floating" or "inline" | | data-position | "bottom-right" | "bottom-right" or "bottom-left" | | data-target | — | CSS selector for inline mode container | | data-color | "#22c55e" | Accent color (FAB and call button) | | data-title | — | Panel header title | | data-logo-url | MrCall logo | Logo image URL | | data-bot-name | — | Subtitle under the title | | data-popup-message | — | Popup bubble text near the FAB | | data-show-popup | "false" | "true" to show popup | | data-show-popup-time | "0" | Seconds before popup appears | | data-auto-open | "false" | "true" to auto-open the panel |

Dynamic user parameters

User details can be set dynamically after the widget is loaded, before the call starts:

window._mrcallWidget.configure({
  username: "user123",
  email: "[email protected]",
  displayName: "Mario Rossi"
});

configure() accepts any subset of: username, email, displayName, authUser, authPassword, businessId, wsUrl, baseUrl, encoding, language.

Programmatic usage

const widget = new MrCallDirectVoiceWidget({
  baseUrl: "https://your-server.com",
  authUser: "API_USER",
  authPassword: "API_PASS",
  businessId: "YOUR_BUSINESS_ID",
  title: "Talk to our AI",
  position: "bottom-right",
  onCallStarted: (sid) => console.log("Call started:", sid),
  onCallEnded: (reason) => console.log("Call ended:", reason),
  onError: (msg) => console.error("Error:", msg),
  onStatusChange: (status) => console.log("Status:", status)
});

// Update user params at runtime
widget.configure({ username: "john", email: "[email protected]" });

// Teardown
widget.destroy();

SDK — Programmatic API

For full control without the UI widget.

Security Architecture

The SDK separates session initialization (requires API credentials) from audio streaming (uses a time-limited access key). This enables two deployment patterns depending on your security requirements.

Pattern A — Backend-initialized (recommended for production)

API credentials stay on your server. The browser never sees them.

 Browser                Your Backend              StarChat
    |                        |                        |
    |  "start call"          |                        |
    |----------------------->|                        |
    |                        |  POST /voice/init/BID  |
    |                        |  Authorization: Basic  |
    |                        |----------------------->|
    |                        |  { sessionId,          |
    |                        |    accessKey, wsUrl }  |
    |                        |<-----------------------|
    |  { wsUrl }             |                        |
    |<-----------------------|                        |
    |                                                 |
    |  WebSocket to wsUrl (no credentials)            |
    |------------------------------------------------>|
    |  <-- binary audio (Opus or PCM16) -->           |
const voice = new MrCallDirectVoice({
  baseUrl: 'https://your-starchat-server.com'
});
voice.onCallStarted = (sid) => console.log('Active:', sid);
await voice.startStream(wsUrl); // wsUrl from your backend

Pattern B — Frontend-initialized (demos, internal tools)

const voice = new MrCallDirectVoice({
  authUser: 'apiUser',
  authPassword: 'apiPass',
  businessId: 'biz-123',
  username: 'john',
  baseUrl: 'https://your-starchat-server.com'
});
await voice.startCall();

Quick Start

1. Include the SDK

<script src="dist/MrCallDirectVoice.bundle.min.js"></script>

Or as an ES module:

<script type="module">
  import MrCallDirectVoice from './MrCallDirectVoice.js';
</script>

AudioWorklet processors and the Opus codec are fully inlined — no extra files to serve.

2. Backend-initialized flow (Pattern A)

Your backend calls the init API:

curl -u apiUser:apiPass -X POST \
  "https://host/mrcall/v1/voice/init/BUSINESS_ID?username=john&sampleRate=24000&encoding=opus"

Your backend passes wsUrl to the frontend:

const voice = new MrCallDirectVoice({ baseUrl: 'https://starchat-host' });

voice.onCallStarted = (sessionId) => console.log('Call active:', sessionId);
voice.onCallEnded   = (reason)    => console.log('Call ended:', reason);
voice.onError       = (msg)       => console.error('Error:', msg);

await voice.startStream(wsUrl);
voice.hangup();

3. Frontend-initialized flow (Pattern B)

const voice = new MrCallDirectVoice({
  authUser: 'apiUser',
  authPassword: 'apiPass',
  businessId: 'your-business-id',
  username: 'end-user-id',
  email: '[email protected]',
  displayName: 'John Doe',
  encoding: 'opus',
  baseUrl: 'https://starchat-host'
});

voice.onCallStarted = () => console.log('Active');
voice.onCallEnded   = () => console.log('Ended');
voice.onError       = (e) => console.error(e);

await voice.startCall();
voice.hangup();

4. Cleanup

window.addEventListener('beforeunload', () => voice.destroy());

API Reference

Constructor

new MrCallDirectVoice(options)

| Option | Type | Required | Default | Description | |----------------|--------|-------------------|----------|------------------------------------------| | baseUrl | string | always | | StarChat server URL (https) | | encoding | string | no | "opus" | "opus" or "pcm16" | | language | string | no | | ISO language tag (e.g. it-IT). Overrides business default language for TTS/STT. | | sampleRate | number | no | 24000 | Audio sample rate: 8000, 16000, or 24000 | | authUser | string | for startCall() | | API username (Basic Auth) | | authPassword | string | for startCall() | | API password (Basic Auth) | | businessId | string | for startCall() | | CRM business identifier | | username | string | for startCall() | | End-user identifier | | email | string | no | | End-user email | | displayName | string | no | | End-user display name |

Methods

| Method | Description | |--------|-------------| | startCall() | Full flow. Calls the init API with credentials, then opens the WebSocket. Returns a Promise. | | startStream(wsUrl) | Stream-only flow. Opens the WebSocket to a pre-obtained wsUrl. No credentials needed. Returns a Promise. | | hangup() | Sends hangup signal and closes the connection. | | destroy() | Force-releases all resources (mic, audio, WebSocket, codec). Safe to call at any time. | | static initSession(opts) | Standalone init call. Returns Promise<{sessionId, accessKey, wsUrl, encoding}>. |

Properties

| Property | Type | Description | |----------------|---------|-------------------------------------------------------------| | status | string | "idle" | "connecting" | "active" | "ending" | | sessionId | string | Server-assigned session ID (available after callStarted) | | inCall | boolean | true when connecting or active | | encoding | string | Effective encoding after detection | | opusSupported | boolean (static) | true if WebCodecs Opus is available | | version | string (static and instance) | SDK version, e.g. "1.0.6" |

Callbacks

| Callback | Signature | When | |------------------|------------------------|---------------------------------------| | onCallStarted | (sessionId) => void | Server confirms call is active | | onCallEnded | (reason) => void | Call terminated (local or remote) | | onError | (message) => void | WebSocket or server error | | onStatusChange | (status) => void | Any status transition | | onAudioStats | (stats) => void | First few audio frames, each direction |

onAudioStats

Diagnostics for the first audioStatsFrames frames per direction (default 5, 0 disables). It answers the first question worth asking when someone reports a dead call: is audio not arriving at all, or arriving silent?

voice.onAudioStats = ({ direction, seq, bytes, samples, peak }) => {
  console.log(`${direction} #${seq}: ${bytes}B`, samples ? `${samples} samples, peak=${peak}` : "");
};
// send #1: 87B
// recv #1: 960B 480 samples, peak=8241

samples and peak are present whenever the PCM is in hand: always on recv, and on send only in pcm16 mode, since an outgoing Opus frame is already encoded. A peak of 0 means real audio arrived and it was silence, which is a different fault from no frames at all. The callback goes quiet after the cap, so it never becomes a per-packet cost for the life of the call.

Status Lifecycle

idle ──> connecting ──> active ──> ending ──> idle
              |                       |
              └── (error) ──> idle    └── (error) ──> idle

Backend Integration Examples

Node.js / Express

app.post('/api/voice/start', async (req, res) => {
  const { username, businessId } = req.body;

  const response = await fetch(
    `${STARCHAT_URL}/mrcall/v1/voice/init/${businessId}?username=${username}&sampleRate=24000&encoding=opus`,
    {
      method: 'POST',
      headers: { 'Authorization': 'Basic ' + btoa(API_USER + ':' + API_PASS) }
    }
  );

  const session = await response.json();
  res.json({ wsUrl: session.wsUrl });
});

Python / Flask

@app.route('/api/voice/start', methods=['POST'])
def start_voice():
    data = request.json
    r = requests.post(
        f'{STARCHAT_URL}/mrcall/v1/voice/init/{data["businessId"]}',
        params={'username': data['username'], 'sampleRate': 24000, 'encoding': 'opus'},
        auth=(API_USER, API_PASS)
    )
    session = r.json()
    return jsonify({'wsUrl': session['wsUrl']})

Init API Reference

POST /mrcall/v1/voice/init/{businessId}

Auth: Basic HTTP (Authorization header).

Query parameters:

| Param | Required | Description | |---------------|----------|------------------------------------| | username | yes | End-user identifier | | sampleRate | no | 8000, 16000, or 24000 (default) | | encoding | no | opus (default) or pcm16 | | email | no | End-user email | | displayName | no | End-user display name | | language | no | ISO language tag (e.g. it-IT). Overrides business default. |

Response (200):

{
  "sessionId": "abc123-...",
  "accessKey": "fa3b2c...128-hex-chars",
  "wsUrl": "/mrcall/v1/voice/stream/abc123-.../fa3b2c...?encoding=opus",
  "encoding": "opus"
}

WebSocket Protocol

URL: wss://{host}/mrcall/v1/voice/stream/{sessionId}/{accessKey}[?encoding=opus]

No authentication headers needed — the access key in the path is the auth.

Binary frames:

  • PCM16: Raw little-endian signed 16-bit integer, mono.
  • Opus: Raw opus-encoded frames (20ms, mono, 24 kbps VoIP).

Text frames (JSON):

| Direction | Type | Fields | Description | |-----------|------------------|---------------------|---------------------------------| | Client | ping | | Keepalive (SDK sends every 5s) | | Server | pong | | Keepalive response | | Client | hangup | | End the call | | Server | callStarted | sessionId | Call is active, start streaming | | Server | callEnded | reason | Call terminated | | Server | error | message | Error occurred | | Server | encodingChanged | encoding | Server fell back to different encoding | | Server | clearAudio | | Barge-in: drop all buffered playback |

clearAudio arrives when the user speaks over the assistant and the server discards its outbound audio queue. The SDK handles it automatically: it drops the frames still queued in the Opus decoder and flushes the playback worklet, keeping a 5ms fade so the stop is not audible as a click. Without it the assistant would keep talking until the local buffer drained, which is what makes barge-in feel sluggish on the web compared to a phone call. A client that ignores the message still works, it just interrupts late.


Build

npm install
npm run build

Output in dist/:

  • MrCallDirectVoice.bundle.js / .min.js — SDK only
  • MrCallDirectVoiceWidget.bundle.js / .min.js — Widget + SDK

Requirements

  • Modern browser with AudioWorklet (Chrome, Edge, Firefox, Safari 14.1+)
  • For Opus: WebCodecs API (Chrome 94+, Edge 94+, Safari 16.4+). Falls back to PCM16 automatically.
  • HTTPS in production (microphone requires secure context). localhost and file:// work for testing.
  • Node.js >= 16 (build only)

Project Structure

MrCallDirectVoice/
  MrCallDirectVoice.js              # SDK source
  MrCallDirectVoiceWidget.js        # Widget source (imports SDK)
  dist/
    MrCallDirectVoice.bundle.js     # SDK — debug
    MrCallDirectVoice.bundle.min.js # SDK — minified + obfuscated
    MrCallDirectVoiceWidget.bundle.js     # Widget — debug
    MrCallDirectVoiceWidget.bundle.min.js # Widget — minified + obfuscated
    mrcall-icon-512x512.png         # Default widget logo
  sample.html                       # SDK interactive demo
  sample-widget.html                # Widget interactive demo
  sample-widget-embed.html          # Widget one-line embed demo
  package.json
  rollup.config.js