Softphone — JsSIP Demo

⚙️ SIP Connection

WebSocket URL (Asterisk WSS)

SIP URI (Asterisk extension)

Authorization username

SIP password

Use the credentials of an internal Asterisk WebRTC extension, not your external SIP provider account. The password is not stored in the browser.

Event Log

📞 Phone

Number or SIP URI

Connect to start calling

📲 Incoming Call

Note: Real calls require HTTPS (or localhost), an Asterisk server configured for WSS/WebRTC, and a SIP route to your provider. Audio uses WebRTC, not Nginx.