Hi Daniel -  See if this configuration helps you.  I was able to make things work using this config as a roadmap

https://github.com/havfo/WEBRTC-to-SIP

On Fri, Feb 12, 2021 at 11:09 AM Daniel Siemens <dwsiemens@msts.com> wrote:

I am trying to get a webrtc setup going.  Here is what I have

 

 

I have  asterisk server at 10.123.244.18.   The webrtc works internally from the freepbx UCP application   As well as the Raspberry phone allocation.   This server doesn’t have any nat on it   all devices are on local / reouted networks. 

 

 

I have a Proxy server  at 10.123.245.30 address.     This server is located in AWS and has an elastic IP. 

 

 

On this server I have ngiinx that will load the  raspberry phone up.     

 

What configuration do I need in  kamailio and rtpengine  to get this working.     

 

If I forward port 8089 in nginx to the /ws side on my asterisk server I can get a call to bridge but with no audio and the call end at 30 seconds when remote.  It works internally fine.    Likely beccuae the  web browser can get to https://10.123.244.18:8089/ws   ports fine. 

 

Thanks.  


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