# Configuring webRTC client SIMPL5

## PBX Configuration <a href="#nastrojka_ats" id="nastrojka_ats"></a>

1. Create a new employee [**account**](/mikopbx/english/manual/telephony/extensions.md).
2. Go to **System → General Settings → SIP** and enable the "**Use WebRTC**" toggle switch.

This automatically creates a dedicated WebRTC endpoint for each internal number (e.g. `204-WS`), allowing simultaneous operation over both PJSIP and WebRTC protocols. No additional per-extension configuration is required.

<figure><img src="/files/0V9IUh97xjiRodU5Vxy7" alt=""><figcaption><p>Switch "Use WebRTC"</p></figcaption></figure>

3. In the [**General Settings**](/mikopbx/english/manual/system/general-settings.md) section, specify the STUN server address. For example, **stun.sipnet.ru**.

<figure><img src="/files/gH6W2PDowDwQ40lECS6p" alt=""><figcaption><p>Stun server</p></figcaption></figure>

4. Verify the WebSocket connection by opening the following URL in your browser:
   * **Local network (no SSL):** `http://PBX_ADDRESS:8088/asterisk/ws`
   * **With SSL certificate:** `https://PBX_ADDRESS:8089/asterisk/ws`

If Asterisk responds, the setup was successful.

> **Note:** For connections over the public internet, a trusted SSL certificate is required so that browsers allow microphone access. We recommend the [Let's Encrypt Module](/mikopbx/english/modules/miko/module-get-ssl-lets-encrypt.md).

## WebRTC Client Setup <a href="#nastrojka_web_rtc_klienta" id="nastrojka_web_rtc_klienta"></a>

1. Open the sipml5 demo in your browser: follow the link "[Enjoy our live demo](https://www.doubango.org/sipml5/call.htm?svn=252)".
2. Fill in the main fields:

<figure><img src="/files/bmipWZjAGlg6k3D3uZg6" alt=""><figcaption></figcaption></figure>

| Field                | Value                                                              |
| -------------------- | ------------------------------------------------------------------ |
| **Display Name**     | Any name                                                           |
| **Private Identity** | `INTERNAL_NUMBER` (e.g. `204`)                                     |
| **Public Identity**  | `sip:INTERNAL_NUMBER-WS@PBX_ADDRESS` (e.g. `sip:204-WS@192.0.2.1`) |
| **Password**         | The extension's SIP password                                       |
| **Realm**            | `PBX_ADDRESS`                                                      |

3. Click **"Expert mode?"** and set the WebSocket Server URL:

<figure><img src="/files/sh2sH9hYMvAFixdmI1aW" alt=""><figcaption></figcaption></figure>

| Connection type        | WebSocket Server URL                     |
| ---------------------- | ---------------------------------------- |
| Local network (no SSL) | `ws://192.0.2.1:8088/asterisk/ws`        |
| With SSL certificate   | `wss://pbx.example.com:8089/asterisk/ws` |

4. Click **Login**. You can now start making calls.


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