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(a)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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