[Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?

[Giovanni Maruzzelli]

Yes, is relevant
Yes, is simple: after a standard install, modify internal.xml SIP profile
and activate port 7443 as wss port, then write dialplan that bridge
incoming calls to outbound calls on your other sip server
You will need "real" ssl certificates, eg letsencrypt, from which you will
build the certificate needed by freeswitch for webrtc:

cat /etc/dehydrated/certs/XXX/fullchain.pem
/etc/dehydrated/certs/XXX/privkey.pem > /etc/freeswitch/tls/wss.pem

That's it

-giovanni


On Wed, Apr 15, 2020 at 12:45 AM <mickael.bride at orange.com> wrote:

> Hello all,
>
>
>
> Sorry in advance if my email is not clear enough or relevant.
>
> My need is to have some kind of WebRTC gateway between a web application
> and a call server managing SIP/RTP (the webapp is a softphone, audio only).
>
> This call server is not exposed on Internet, this is mainly why I need a
> proxy/gateway.
>
>
>
> I am quite a newbie on WebRTC / SIP / RTP.
>
> From what I have seen on Internet some projects/products are doing that
> kind of functionality (Asterisk, Doubango webrtc2sip, Janus, Audio Code …)
> , but I am not able to focus on the simplest way to do it. Also some are
> commercial products but I would prefer open source, installed on CentOS.
>
> FreeSwitch seems really great but I wonder if it is not too “big” or
> complicated for my needs.
>
>
>
> Questions are:
>
> ·         Is FreeSwitch relevant to have a proxy/gateway between a WebRTC
> web app and a call server not exposed on Internet?
>
> ·         Is it quite “simple” to install and configure freeSwitch to
> have that functionality?
>
> ·         Is there any other simpler project to do this?
>
>
>
> Any advice would be really appreciate.
>
>
>
> Thank you,
>
>
>
> Mickaël
>
>
>
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-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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