Setting up Webphone (developed by Mizutech and formerly known as Mizu Webphone) allows you to deploy a fully functional browser-based SIP client. It integrates directly with VoIP servers (like Asterisk, FreePBX, or Mizu Softswitch) using modern WebRTC and JavaScript APIs. Step 1: Download and Deploy the Package
Download the Package: Get the complete zip archive from the Mizutech Webphone Download Portal. This contains the JavaScript API, default skins, and pre-built examples. Extract Files: Unzip the contents on your local system.
Upload to Web Server: Move the entire webphone/ folder to your production web server.
Security Note: While you can run a local test file, production setups require an HTTPS-enabled web host to access device microphones. Step 2: Configure Server and Account Connections
Open the webphone_config.js or webphone_api.js file located in the root of the webphone directory. You need to hardcode or dynamically pass your SIP credentials. Modify the parameters block within the script: javascript
var parameters = { // The explicit URL where your webphone folder is hosted webphonebasedir: ‘https://yourdomain.com’, // Your VoIP/SIP Server domain name or IP address serveraddress: ‘sip.yourvoipserver.com’, // Optional preset credentials (leave blank if your login page handles this) username: ‘SIP_EXTENSION_USER’, password: ‘SIP_EXTENSION_PASSWORD’, // Recommended logging level for early troubleshooting (1-5) loglevel: 5 }; Use code with caution.
(Note: You can reference the full parameter lists via the documentation provided in your download package). Step 3: Launch or Embed the Phone UI
Mizutech includes pre-built HTML files you can launch instantly or frame into your application: WebPhone -SIP for browsers – Mizu VoIP
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