> ## Documentation Index
> Fetch the complete documentation index at: https://docs.phone.wixzel.com/llms.txt
> Use this file to discover all available pages before exploring further.
# Realtime: web and mobile
> Put an agent in a web page or an app over a WebSocket. No SIP trunk, no phone number.
A phone call is one way to reach an agent. Realtime is the other: your user
talks to the same agent from a browser tab, a mobile app or a kiosk, over a
WebSocket. The engines, prompts, knowledge bases and appointment booking are
the ones your phone calls use — only the line is different.
Two steps, and the split is the security model:
1. **Your server** calls `POST /v1/realtime/sessions` with your API key and gets
a `client_secret`.
2. **Your client** opens `wss://api.phone.wixzel.com/v1/realtime?client_secret=…`
and streams audio.
The secret works **once**, for **one minute**, for **one agent**. Your API key
never leaves your server.
## 1. Create a session (server)
```bash cURL theme={null}
curl -X POST https://api.phone.wixzel.com/v1/realtime/sessions \
-H "Authorization: Bearer $WIXZEL_API_KEY" \
-H "Content-Type: application/json" \
-d '{
"agent_id": "6a96a3ead6e886d42462dd3e",
"allowed_origins": ["https://app.example.com"],
"metadata": { "user_id": "u_812" }
}'
```
```ts TypeScript theme={null}
const session = await wixzel.realtime.createSession({
agent_id: '6a96a3ead6e886d42462dd3e',
allowed_origins: ['https://app.example.com'],
metadata: { user_id: 'u_812' },
});
// send { url: session.url, client_secret: session.client_secret } to your client
```
```dart Dart theme={null}
final session = await client.realtime.createSession(CreateRealtimeSession(
agentId: '6a96a3ead6e886d42462dd3e',
metadata: {'user_id': 'u_812'},
));
```
```json Response theme={null}
{
"id": "rt-5b0c7c0e-9d1f-4c52-8a9e-2f0d3a4b6c7d",
"object": "realtime_session",
"client_secret": { "value": "rt_secret_…", "expires_at": "2026-09-12T10:01:00.000Z" },
"url": "wss://api.phone.wixzel.com/v1/realtime",
"agent_id": "6a96a3ead6e886d42462dd3e",
"engine": "gemini_live",
"max_duration_seconds": 600,
"audio_format": "mulaw_8000"
}
```
| Field | |
| ---------------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
| `agent_id` | Required. One of your agents. |
| `lead_id` | Optional. Attributes the session to a lead: `{{name}}` merge fields resolve from it and appointment booking uses its phone. Without one, merge fields resolve to nothing. |
| `metadata` | Optional. Your own ids, echoed on the resulting call and its `callCompleted` webhook. Never sent to the model. |
| `max_duration_seconds` | Optional. 10–3600, default 600. The server ends the session at this point. |
| `allowed_origins` | Optional, recommended for browsers. The socket is refused unless the page's `Origin` matches, so a secret copied out of your page cannot be used from another site. |
Needs the `calls:write` scope. **Nothing is charged here** — a user who opens
your page and then denies the microphone costs you nothing.
## 2. Connect (client)
### In a browser
The TypeScript SDK ships a browser client that handles the microphone,
encoding, playback and barge-in:
```ts theme={null}
import { RealtimeSession } from 'wixzel-phone/realtime';
// `session` is what your server returned above
const call = await RealtimeSession.connect(session, {
onTranscript: ({ role, text }) => render(role, text),
onAgentSpeaking: (speaking) => setSpeaking(speaking),
onError: ({ code, message }) => showError(code, message),
onEnd: (reason) => showEnded(reason),
});
// later
call.setMuted(true);
await call.end();
```
It needs a secure page (`https://`, or `localhost`) and asks for the
microphone when `connect` is called. To ask **before** you mint a session,
call `requestMicrophone()` first and pass the stream to `session.start()`.
### In Flutter or Dart
The Dart SDK has the protocol client. Microphone and speaker access are your
app's — use any recording plugin that gives you 16-bit PCM at 8 kHz, and any
PCM player:
```dart theme={null}
import 'package:wixzel_phone/wixzel_phone.dart';
final conn = RealtimeConnection.forSession(session);
conn.events.listen((e) {
switch (e) {
case RealtimeAudio(:final mulaw): player.add(mulawToPcm16(mulaw));
case RealtimeAudioClear(): player.clear(); // the user interrupted
case RealtimeTranscript(:final role, :final text): print('$role: $text');
case RealtimeError(:final code, :final message): print('$code: $message');
case RealtimeEnded(:final reason): print('ended: $reason');
default:
}
});
mic.listen((Int16List pcm) => conn.sendAudio(pcm16ToMulaw(pcm)));
// later
await conn.end();
```
Turn on your platform's echo cancellation. Without it the agent hears itself
through the speaker and interrupts itself.
### Anything else
The protocol is plain JSON over a WebSocket, below.
## The protocol
Every frame is a JSON text message with a `type`. Audio is **base64 G.711
µ-law, 8 kHz, mono** in both directions — exactly what a phone line carries.
Send it in 20 ms frames (160 bytes).
**From the server**
| `type` | Fields | Meaning |
| ----------------- | ------------------------------------------------------------------------------------- | --------------------------------------------------------------------------------- |
| `session.started` | `session_id`, `call_id`, `agent_id`, `engine`, `max_duration_seconds`, `audio_format` | Admitted; the agent is connecting and will greet the user. |
| `audio` | `audio` | Agent speech. Play it in order. |
| `audio.clear` | | The user interrupted. Drop everything you have queued. |
| `transcript` | `role` (`user` or `assistant`), `text` | A line of the conversation. |
| `session.warning` | `code`, `seconds_remaining` | `low_balance`: the account is about to run out of credit. |
| `error` | `code`, `message` | Something went wrong. Before `session.started`, it means the session was refused. |
| `session.ended` | `reason`, `duration_seconds` | Over. The socket closes next. |
**From the client**
| `type` | Fields | Meaning |
| ------------- | ------- | ----------------- |
| `audio` | `audio` | Microphone audio. |
| `session.end` | | Hang up. |
While the agent is saying its opening line, keep sending audio but send
silence (`0xFF` bytes): the agent then cannot be interrupted by room noise
before the user has heard anything. The browser client does this for you.
### Why a session ends
| `reason` | |
| ---------------------- | --------------------------------------------------------- |
| `client` | You sent `session.end`. |
| `agent` | The agent ended the conversation — the user said goodbye. |
| `max_duration` | It reached `max_duration_seconds`. |
| `idle` | Nobody spoke for two minutes. |
| `insufficient_credits` | The account ran out of credit mid-session. |
| `error` | The voice engine failed. |
| `disconnected` | The socket dropped. |
### Refusals
A browser cannot read the HTTP status of a refused WebSocket, so refusals
arrive as an `error` frame followed by a close with one of these codes:
| Close | `code` | Fix |
| ----- | ------------------------------------------ | ------------------------------------------------------------------------------------------------------- |
| 4401 | `invalid_client_secret`, `invalid_api_key` | The secret is missing, expired or already used — mint a new one. Or the key that minted it was revoked. |
| 4402 | `insufficient_credits` | Top up. |
| 4403 | `origin_not_allowed` | The page's origin is not in `allowed_origins`. |
| 4404 | `agent_not_found` | The agent was deleted after minting. |
| 4409 | `concurrency_limit` | The account is at its concurrent-call limit. |
| 4429 | `spend_limit` | The API key reached its spend limit. |
| 4503 | `engine_unavailable`, `server_busy`, … | Try again shortly. |
## Billing and records
A realtime session **is a call** to everything but the phone network:
* It is billed at the **same per-minute price** as a phone call on the same
engine, per second, from credit. See [pricing](https://phone.wixzel.com/pricing).
* It counts toward your **concurrent-call limit** (5 by default) and the API
key's **spend limit**.
* It appears in `GET /v1/calls` with `"channel": "web"` (filter with
`?channel=web`), inbound, with no `from` or `to`, and with its transcript.
* It ends with a `callCompleted` [webhook](/webhooks) carrying
`"provider": "web"` and your `metadata`.
Human transfer is not available: there is no second line to hand the user to.
Web sessions are not recorded.