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

[Giovanni Maruzzelli]

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On Wed, Apr 15, 2020 at 5:38 PM Yan Zhang <yan at petatrading.com> wrote:

> WHY I AM STILL RECEIVING THIS ENDLESS MAIL NOTICE?
>
> Please remove me. Thanks
>
>
>
> On Tuesday, April 14, 2020, 06:48:26 PM EDT, <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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