<div dir="ltr"><div>Yes, is relevant</div><div>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</div><div>You will need "real" ssl certificates, eg letsencrypt, from which you will build the certificate needed by freeswitch for webrtc:</div><div><pre><code>cat /etc/dehydrated/certs/XXX/fullchain.pem /etc/dehydrated/certs/XXX/privkey.pem > /etc/freeswitch/tls/wss.pem<br><font size="2"><span style="font-family:arial,sans-serif"><br></span></font></code></pre><pre><code><font size="2"><span style="font-family:arial,sans-serif">That's it</span></font><br><br></code></pre><pre><code>-giovanni<br><br></code></pre></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Wed, Apr 15, 2020 at 12:45 AM <<a href="mailto:mickael.bride@orange.com">mickael.bride@orange.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">
<div lang="FR">
<div class="gmail-m_-4416458266192410951WordSection1">
<p class="MsoNormal">Hello all,<u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal"><span lang="EN-US">Sorry in advance if my email is not clear enough or relevant.<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">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).<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">This call server is not exposed on Internet, this is mainly why I need a proxy/gateway.<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">I am quite a newbie on WebRTC / SIP / RTP.
<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">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.<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">FreeSwitch seems really great but I wonder if it is not too “big” or complicated for my needs.<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">Questions are: <u></u><u></u></span></p>
<p class="gmail-m_-4416458266192410951MsoListParagraph"><u></u><span style="font-family:Symbol" lang="EN-US"><span>·<span style="font:7pt "Times New Roman"">
</span></span></span><u></u><span lang="EN-US">Is FreeSwitch relevant to have a proxy/gateway between a WebRTC web app and a call server not exposed on Internet?<u></u><u></u></span></p>
<p class="gmail-m_-4416458266192410951MsoListParagraph"><u></u><span style="font-family:Symbol" lang="EN-US"><span>·<span style="font:7pt "Times New Roman"">
</span></span></span><u></u><span lang="EN-US">Is it quite “simple” to install and configure freeSwitch to have that functionality?<u></u><u></u></span></p>
<p class="gmail-m_-4416458266192410951MsoListParagraph"><u></u><span style="font-family:Symbol" lang="EN-US"><span>·<span style="font:7pt "Times New Roman"">
</span></span></span><u></u><span lang="EN-US">Is there any other simpler project to do this?
<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">Any advice would be really appreciate.<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-US">Thank you,<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-US"><u></u> <u></u></span></p>
<p class="MsoNormal">Mickaël<span><u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
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<a href="https://freeswitch.com" rel="noreferrer" target="_blank">https://freeswitch.com</a></blockquote></div><br clear="all"><br>-- <br><div dir="ltr" class="gmail_signature">Sincerely,<br><br>Giovanni Maruzzelli<br>OpenTelecom.IT<br>cell: +39 347 266 56 18<br><br></div>