I'm sorry, it looks like I missed the part where you mentioned that you have it all working now, that's why I was asking about where you router is because it very much sounded like routing/nat issues you were having :)<div><br></div><div>Paul<br><br>On Thu, 1 Aug, 2013 at 6:34 PM, Peter <eidevm5@gmail.com> wrote:<br>
<blockquote type="cite"><div dir="ltr">Correct. Both Kamailio servers are on different networks. I didn't show the router in the mix for simplicity.<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Aug 2, 2013 at 6:31 AM, Paul <span dir="ltr"><<a href="mailto:pasha@prosperity4ever.com" target="_blank">pasha@prosperity4ever.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Where on your diagram is the Router? I assume that your Kamailio 1 and Kamailio 2 are on two different networks?<span class="HOEnZb"><font color="#888888"><div><br></div></font></span><div><span class="HOEnZb"><font color="#888888">Paul</font></span><div>
<div class="h5"><br><br>On Wed, 31 Jul, 2013 at 10:12 PM, Peter <<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>> wrote:<br>
<blockquote type="cite"><div dir="ltr">After checking everything for the umpteenth time, I discovered I was missing a route back to <a href="http://10.1.254.0/24" target="_blank">10.1.254.0/24</a><br><br>Once I put that in, bingo, audio!<br>
</div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Thu, Aug 1, 2013 at 12:33 PM, Peter <span dir="ltr"><<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div><div><div><div><div><div><div><div><div><div><div>I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end.<br><br></div>The set up is as follows:<br>
<br><br><br></div>Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000)<br><br><br></div>Using Freeswitch 1.2.12 on CentOS (installed via RPM)<br><br></div>
Freeswitch has two interfaces:<br><br></div>external - 10.1.1.206<br></div>internal - 10.10.10.206<br><br></div>Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch.<br>
<br></div>The SIP negotiation is working as I can call either Linphone client. <br><br></div>I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch.<br>
<br></div>I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio. <br><br></div>Any pointers on how to track down the issue?<br><br>Thanks<span><font color="#888888"><br>
<br>Peter<br><br>
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