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<div style="direction: ltr;font-family: Tahoma;color: #000000;font-size: 10pt;">Hi Matt,
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The SDP is being generated by freeswitch and you would need to make it think that when its sending to kamailio, that kamailio is an external host so freeswitch uses its public IP address in the SDP that it is then forwarded on directly to the carrier.</div>
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<div>I've never used freeswitch but thats roughly what you'd do with asterisk.</div>
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Thanks</div>
<div>Kenny Watson</div>
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<div id="divRpF558406" style="direction: ltr;"><font face="Tahoma" size="2" color="#000000"><b>From:</b> users-bounces@lists.opensips.org [users-bounces@lists.opensips.org] on behalf of matt [matt@supportedbusiness.com]<br>
<b>Sent:</b> 22 October 2014 08:42<br>
<b>To:</b> users@lists.opensips.org<br>
<b>Subject:</b> [OpenSIPS-Users] Load balancer setup<br>
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<div dir="ltr">Hi,
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<div>I was looking for some guidance on using the load balancer in a NAT environment.</div>
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<div>I have the following setup (the IP addresses are made up but should give an indication):</div>
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<div>1 x opensips server with load balancer module - IP 192.168.0.1<br>
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<div>2 x freeswitch servers - IP 192.168.0.2 & 192.168.0.3</div>
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<div>All 3 servers have seperate external IP address routing to their internal IP via our firewall:</div>
<div>217.0.0.1 routed to 192.168.0.1 (Opensips)</div>
<div>217.0.0.2 routed to 192.168.0.2 (FS1)<br>
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<div>217.0.0.3 routed to 192.168.0.3 (FS2)<br>
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<div>I have the load_balancer table with the following details:</div>
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<div>id, | group_id, | dst_uri, | resources, | probe_mode, | description</div>
<div>'1', | '1', | 'sip:<a href="http://192.168.0.2:5080" target="_blank">192.168.0.2:5080</a>', | 'pstn=10', | '2', | 'FS1'</div>
<div>'2', | '1', | 'sip:<a href="http://192.168.0.3:5080" target="_blank">192.168.0.3:5080</a>', | 'vm=1', | '2', | 'FS2'</div>
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<div>The call flow is:</div>
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<div>SIP Provider --> 217.0.0.1 Opensips --> <a href="http://192.168.0.2/3" target="_blank">
192.168.0.2/3</a></div>
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<div>The issue is, that when the 200 ok response is sent to the SIP provider, the Freeswitch server's internal IP is being sent in the SDP connection information (c). This causes the ACK response from the SIP Provider to fail to be sent correctly.</div>
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<div>With the calls routed directly to the FS servers (removing opensips from the flow), the calls work fine.</div>
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<div>Any help would be much appreciated :)</div>
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<div>thanks</div>
<div>Matt</div>
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