WebRTC (SipML5) on Doubango registers but media fails

Asterisk 13.6 - Chrome 48.0.2564.48 beta-m (64-bit)

Following the instructions at wiki.asterisk.org/wiki/display/ … ing+SIPML5, somehow I managed to get the Douabango SipML5 demo webpage to register and login to my server. But when it comes to making a call, I get the following errors:

In the browser console, it seems to be trying…

GET http://www.doubango.org/sipml5/null Request Method:GET Status Code:404 Not Found
and by the dialler, the following message flashes up

One thing to note - I had to change https://www.doubango.org/sipml5/call.htm to http://www.doubango.org/sipml5/call.htm otherwise it was complaining about me attempting an insecure call. I might be being thick here, but I can’t find how to configure the wss 8089 port.

That said, once I changed to http, it definitely registers just fine on my Asterisk.

Before I keep on too late into the night, does that “404 not found” error look like it might be something at their end? I don’t want to start trying to debug a problem that’s somewhere else.

By the way, when the call is attempted, nothing flashes by in the pjsip log, so it’s not even hitting my server, unlike the registration which I can see going just fine :smile:

“Media stream permission denied” is because

One major change is WebRTC won’t work without HTTPS.
According internal security policy Chrome browser does not support getUserMedia() for unsecure pages since version 47. So you will not be able to use microphone if your page is not HTTPS.

flashphoner.com/getusermedia-no- … e-origins/