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 <[email protected]> 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. > > ------------------------------------------------------------------ > This email is intended solely for the use of the > addressee and may contain information that is > confidential, proprietary, or both. If you receive > this email in error please immediately notify > the sender and delete the email. > ------------------------------------------------------------------ > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >
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