WebRTC testing
WebRTC to SIP WebSocket tester
Connect to your PBX WSS endpoint from the browser using SIP.js. Test REGISTER, inspect transport state, and debug WebRTC-to-SIP integration.
Credentials stay in your browser and are sent only to your WSS endpoint. Never test production trunks on shared machines.
Connection log
DisconnectedLogs appear here when you connect…
How it works
Three steps to results
- 01
Enter WSS details
Provide your WebSocket URL (wss://), SIP extension, domain, and password. Credentials stay in your browser.
- 02
Connect and register
The tool opens a WSS transport and sends a SIP REGISTER. Watch transport and registerer state changes in the live log.
- 03
Debug failures
Connection errors, TLS issues, and authentication failures appear in the log with timestamps for quick troubleshooting.
FAQ
Frequently asked questions
Common questions about this tool and how it fits into your VoIP security workflow.
Contact support →Enter your Asterisk WSS URL (e.g. wss://pbx.example.com:8089/ws), extension, domain, and password. Click Connect to attempt a SIP REGISTER via WebSocket.
WebSocket secure?
Port 5060 is probably still exposed to scanners.
See who's attacking your SIP server right now in our live honeypot threat feed.