[SR-Users] basic help, In over my head in getting a webrtc up.

[Jeremy McNamara]

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 at msts.com> wrote:

> I am trying to get a webrtc setup going.  Here is what I have
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> 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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> 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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> On this server I have ngiinx that will load the  raspberry phone up.
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> What configuration do I need in  kamailio and rtpengine  to get this
> working.
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> 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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