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.
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.
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.