Hi,

I've been testing your webphone solution, and it works perfectly with your online webrtc2sip gateway. Now i've tried to implement the gateway on my own server, and while i get the webphone to connect to my VoIP server (3cx), i'm getting no audio on phone calls. Also, when i disconnect the call on the other end, the webphone reports it's still in a call. Any ideas on what i might be doing wrong?
Everything is set up on a local lan, and firewall is disabled on webrtc2sip server.

Another thing i noticed:
C:\Users\Administrator>netstat -an | findstr :506
  TCP    192.168.30.26:5060     0.0.0.0:0              LISTENING
  TCP    192.168.30.26:5061     0.0.0.0:0              LISTENING
  UDP    192.168.30.26:5060     *:*

It looks like it's not listening on UDP. Any idea why?

I've tested sipml5, and i'm getting exactly the same behaviour as with your webphone. It connects to the server, starts a call, my phone rings, but there's no audio when i answer, and when i hang up on the phone softphone says it's still in call.

I've received some new public ip's today so i'll try to move entire testing solution there and see how it goes. In the meantime, i'm guessing i've set up something wrong in webrtc gateway.

I'm not using 3cx webrtc. It doesnt actuqally work for what i need it to. I have allready installed your free WebRTC gateway and that's the one i'm having problems with.

I just want to add here that now this issue is fixed also in the public downloadable package: free webrtc sip gateway