Test numbers for SIP and WebRTC


Want to test or troubleshoot WebRTC calling or SIP calling without bothering anybody?

Now there are some test numbers you can call by just clicking in your desktop or mobile web browser.

This can give you instant feedback about whether your SIP device, sound drivers, softphone and/or browser are in a good state. If using WebRTC, you can make calls to this service while studying the JavaScript console logs in your browser to see exactly how the protocol works.

I hope this well help further increase the rate of positive outcomes for RTC projects using free software and open standards. Please feel free to provide any feedback about this service.