Configuring webRTC client SIMPL5

PBX Configuration

1. Create a new employee account.

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.

Switch "Use WebRTC"

1. In the General Settings section, specify the STUN server address. For example, stun.sipnet.ru.

Stun server

1. 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.

WebRTC Client Setup

1. Open the sipml5 demo in your browser: follow the link "Enjoy our live demo".

2. Fill in the main fields:

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

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

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

1. Click Login. You can now start making calls.