<div dir="ltr"><div>Thanks Ken.<br><br></div>Any guides on how to configure FS for this purpose? I've tried following the SBC guide on the Wiki, but couldn't work out how to adapt it to my environment.<br><br>Peter<br>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jul 30, 2013 at 1:19 AM, Ken Rice <span dir="ltr"><<a href="mailto:krice@freeswitch.org" target="_blank">krice@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<font face="Monaco, Courier New"><span style="font-size:11pt">Simple answer yes, FS can connect an SRTP call to a non-SRTP call... <br><div><div class="h5">
<br>
<br>
On 7/29/13 2:04 AM, "Ei Dev" <<a href="http://eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>> wrote:<br>
<br>
</div></div></span></font><blockquote><font face="Monaco, Courier New"><span style="font-size:11pt"><div><div class="h5">My scenario is the following:<br>
<br>
External clients register to a Kamailio server (doesn't have to be Kamailio, just what I'm using in my test environment) via TLS and calls between external clients are via SRTP.<br>
<br>
There is another internal Kamailio server where users register via standard SIP and media over RTP.<br>
<br>
<br>
Can Freeswitch act as a gateway for calls between internal and external clients and vice versa, where the SRTP stream is initiated/terminated (depending on the call direction) on the Freeswitch server?<br>
<br>
Thanks.<br>
<br>
<br>
<br>
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