Ok, nice to hear it. Many thanks to you guys!<div>Another question (maybe dumb) about this setup.š</div><div>Is it necessary to freeswitch to handle registrations or FS can forward them to VoIP provider?š</div><div>I mean, for example, user A and user B have sip accounts at voip provider and they can register at voip provider through fs.š</div>
<div>Or I have to register them in fs independently?<br><br><div class="gmail_quote">2012/9/27 Daniel Ivanov <span dir="ltr"><<a href="mailto:sertys@gmail.com" target="_blank">sertys@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
It is possible, just set proxy_media to false and have fs process the streams and you get independent legs. Your a-leg can talk SRTP, but it's not necessary that the b-leg talks it. SRTP is user-to-system protection scheme, not user-to-user.<br>
<br><div class="gmail_quote"><div><div class="h5">On Thu, Sep 27, 2012 at 4:32 PM, ánton ôeihrib <span dir="ltr"><<a href="mailto:teihrib@gmail.com" target="_blank">teihrib@gmail.com</a>></span> wrote:<br></div></div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">
Hi all!<br>In my case all sip clients uses SRTP, but VoIP provider accepts only RTP packets. I'm thinking about <br>using freeswitch between them, like so:<br>client <=SRTP=> freeswitch <=RTP=> VoIP provider. <br>
Is it possible? If so please give me some direction how to implement this setup.<br>Thanks a lot in advance.
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