<div dir="ltr">Hi Daniel - See if this configuration helps you. I was able to make things work using this config as a roadmap<div><br></div><div><a href="https://github.com/havfo/WEBRTC-to-SIP">https://github.com/havfo/WEBRTC-to-SIP</a><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Feb 12, 2021 at 11:09 AM Daniel Siemens <<a href="mailto:dwsiemens@msts.com">dwsiemens@msts.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
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<p class="MsoNormal">I am trying to get a webrtc setup going. Here is what I have
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<p class="MsoNormal">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.
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<p class="MsoNormal">I have a Proxy server at 10.123.245.30 address. This server is located in AWS and has an elastic IP.
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<p class="MsoNormal">On this server I have ngiinx that will load the raspberry phone up. <u></u><u></u></p>
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<p class="MsoNormal">What configuration do I need in kamailio and rtpengine to get this working. <u></u><u></u></p>
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<p class="MsoNormal">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 <a href="https://10.123.244.18:8089/ws" target="_blank">https://10.123.244.18:8089/ws</a>
ports fine. <u></u><u></u></p>
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<p class="MsoNormal">Thanks. <u></u><u></u></p>
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