<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div style="-webkit-text-size-adjust: auto;">With a patched version of rtpproxy you can advertise your private ip. </div><div style="-webkit-text-size-adjust: auto;"><br></div><div><span style="-webkit-text-size-adjust: auto;"><a href="http://www.fredposner.com/voip/1457/kamailio-behind-nat/">http://www.fredposner.com/voip/1457/kamailio-behind-nat/</a></span><br><br><span style="-webkit-text-size-adjust: auto;">---Fred </span><div style="-webkit-text-size-adjust: auto;"><br></div></div><div style="-webkit-text-size-adjust: auto;"><br>On Jan 21, 2014, at 6:18 AM, "John Smith" <<a href="mailto:jsmith.15@mail.com">jsmith.15@mail.com</a>> wrote:<br><br></div><blockquote type="cite" style="-webkit-text-size-adjust: auto;"><div><span style="font-family:Verdana"><span style="font-size:12px"><p style="margin:0px; padding:0px; font-family: Helvetica;">Hello,</p><p style="margin:0px; padding:0px; font-family: Helvetica; min-height: 14px;"> </p><p style="margin:0px; padding:0px; font-family: Helvetica;">I am currently deploying one Kamailio behind NAT with one Asterisk as explained in the Asipto KB (<a href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb">Kamailio 4.0.x and Asterisk 11.3.0 using Asterisk Database</a>). The structure is deployed as described in that document, with the only addition of one NAT between Kamailio and Internet:</p><p style="margin:0px; padding:0px; font-family: Helvetica; min-height: 14px;"> </p><p style="margin:0px; padding:0px; font-family: Helvetica;">Phone ———————> Nat ———————> Kamailio ——————> Asterisk</p><p style="margin:0px; padding:0px; font-family: Helvetica; min-height: 14px;"> </p><p style="margin:0px; padding:0px; font-family: Helvetica;">I have declared the private IP with the advertise option in order to support the NAT, enabled WITH_NAT and I have installed rtpproxy using standard Debian package configured as <span style="background-color: #fbfaf9">rtpproxy -l public_ip_ -s udp:localhost:7722</span></p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249); min-height: 14px;"> </p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249);">After setting up two phones which register correctly at Asterisk, I have no audio at all. </p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249); min-height: 14px;"> </p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249);">By placing tcpdumps between nodes I see at Kamailio node both audio from public IP to internal Kamailio IP and from the latter to the Asterisk IP. In Asterisk I see audio coming from the Kamailio private IP and then back to the public IP of the phone.</p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249); min-height: 14px;"> </p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249);">My guess is that audio should flow back into Kamailio and then to the phone, not directly from Asterisk as it is right now.</p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249); min-height: 14px;"> </p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249);">Can anyone hint at where I am wrong?</p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249); min-height: 14px;"> </p><p style="margin:0px; padding:0px; font-family: Helvetica; background-color: rgb(251, 250, 249);">Thank you</p></span></span>
</div></blockquote><blockquote type="cite" style="-webkit-text-size-adjust: auto;"><div><span>_______________________________________________</span><br><span>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list</span><br><span><a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a></span><br><span><a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a></span><br></div></blockquote></body></html>