WebRTC with Kamailio

I noticed lots of queries about this subject, and I created a Kamailio sample script that could help those who are in trouble when working on this.

This post is planned to be explanatory, with some level of detail, but I don’t really have time nowadays so I’m gonna throw it in its current state, hoping to find the time to do it properly soon enough.

  • This script is configured to run behind NAT, port TCP 10080 and TCP/UDP 5090 are exposed to the Internet
  • You have to create valid users using, preferably, “kamctl add …”
  • RTP ports should be open in range 30k-35k, inclusive
  • I used jssip as webrtc SIP UA: http://tryit.jssip.net/
  • Always disable video before placing a call from jssip UA
  • Always use Chrome. Firefox won’t work, and I’m not sure about other browsers
  • I tested calls between:
    • jssip to csipsimple
    • csipsimple to jssip
    • csipsimple to csipsimple

This is the link to the script

Click to Call application using webrtc2sip + asterisk

I’ve been working on this for the last few days and I pleased to say that I managed to get through the series of problems that the learning curve entails and now the app is finally working.

My intention was to learn the fundamentals of webRTC and SIP over websockets and I haven’t found a better solution than the one offered by Doubango Telecom. Their impressive job is transforming the way we communicate and I want to be part of that transformation when it finally hit the “standard technology” label.

The purpose of the application is to demonstrate a new way of doing a click to call service. There are a variety of similar solutions with different approaches being the Java applet the most commonly used amongst them.

Well, although this app doesn’t bring any new thing into the world, it certainly serves the purpose of demonstrating a new way of making things. It’s entirely made using HTML5 with the javascript library that made SIPML5 possible.

The app connects to my lab’s Asterisk, via webrtc2sip which deals with the SIP over WS on one side, and SIP over UDP in the other part. The media is also handled by webrtc2sip by translating the SDP profiles and making transcoding on demand.

You can try it here. It is necessary the latest version of Chrome stable or Firefox Nightly. If you don’t meet the minimum needs, the page will just stay there doing nothing. In case you get an error message it’s probably because my server crashed. Please let me know if that happen.

You can browse the javascript code to see how simple it is. I’m planning to turn the project into something more elaborated and then publish the code. By now, it’s too simple to bother 😉


For some reason, behind very restrictive firewalls the audio is not working properly. I’m working on this to fix it ASAP. Please let me know if that’s your case.


I temporarily deactivated the demo because my server suffered large amounts of hacking attempts. I knew this was possible but I wasn’t expecting this volume.