simple-sip-socket-softphone

WebSocket SIP Softphone

A standalone WebRTC SIP softphone application using SIP.js and WebSocket transport.

https://yannickmcosta.github.io/simple-sip-socket-softphone/softphone.html

Features

• WebRTC-based SIP calling
• WebSocket transport for firewall-friendly connectivity
• Full-featured user interface
• Connection diagnostics and testing
• Real-time call status and duration
• Call history tracking
• DTMF tone support
• Hold/Resume functionality
• Mute controls
• Volume controls
• Incoming call notifications
• Debug logging
• Persistent settings (localStorage)

Quick Start

1. Open softphone.html in a modern web browser
2. Configure your connection settings:
   • WebSocket Server: Your SIP server WebSocket URL (e.g., wss://your-server.example.com:8089/ws)
   • SIP User: Your extension number
   • Password: Your SIP password
   • Domain: Your SIP domain
3. Click Connect to register with the server
4. Enter a phone number and click Call

Server Requirements

This softphone requires a SIP server with WebSocket support, such as:

• Asterisk with chan_sip or chan_pjsip and WebSocket transport
• FreeSWITCH with WebSocket module
• Other SIP servers supporting WebSocket (RFC 7118)

Example Asterisk Configuration

[transport-ws]
type=transport
protocol=ws
bind=0.0.0.0:8089

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0:8089
cert_file=/path/to/cert.pem
priv_key_file=/path/to/key.pem

Browser Support

• Chrome/Chromium 74+
• Firefox 66+
• Safari 12+
• Edge 79+

WebRTC support is required.

Security Considerations

• Always use wss:// (secure WebSocket) in production
• Implement proper authentication on your SIP server
• Consider implementing SRTP for media encryption
• Validate and sanitize all user inputs
• Use HTTPS for hosting the application

License

This project uses SIP.js library (various licenses apply). Check sip-0.21.2.js for details.

Troubleshooting

Cannot Connect

1. Verify WebSocket server is accessible
2. Check firewall rules
3. Ensure correct port (typically 8089 or 8088)
4. For wss://, verify SSL certificate validity

Registration Fails

1. Verify credentials (username/password/domain)
2. Check SIP server logs
3. Ensure user is configured on the server
4. Verify authentication realm matches

No Audio

1. Grant microphone permissions in browser
2. Check audio device selection
3. Verify STUN/TURN configuration if behind NAT
4. Check firewall allows RTP/RTCP packets

Contributing

Contributions are welcome. Please ensure:

• Code follows existing style
• Test changes across multiple browsers
• Update documentation as needed