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

[mickael.bride at orange.com]

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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