<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;">Try using stun server ex<br><br>stun.ekiga.org<br><br>best regards<br><br><br>--- On <b>Thu, 1/5/12, Michal Zubáč <i><michal.zubac@comgate.cz></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"><br>From: Michal Zubáč <michal.zubac@comgate.cz><br>Subject: [Freeswitch-users] bridging 2 SIP destinations from PBX behind NAT<br>To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org><br>Date: Thursday, January 5, 2012, 2:20 PM<br><br><div class="plainMail">Hi.<br><br>I encountered problem with bridging two SIP calls.<br><br>My FreeSwitch is behind NAT, I'm not sure if endpoint switches are, but <br>I think it doesn't matter now.<br>Situation looks like this:<br> SIP1 <--> NAT <--> PBX <--> NAT <--> SIP2<br>SIP1 is caller switch,
SIP2 is destination switch so call is initiated <br>like this:<br> SIP1->PBX->SIP2<br><br>I tested both endpoints separately using ISDN endpoint and they worked.<br>Like this<br> ISDN1->PBX->SIP2<br>&<br> SIP1->PBX->ISDN2<br><br>But when I connected SIP1 & SIP2 together, call is estabilished, but no <br>audio was going through our firewall. Both audio streams from SIP1 and <br>SIP2 were filtered out by our firewall, because Freeswitch (PBX) wasn't <br>sending any initial packets to SIP endpoints, so no NAT holes were created.<br><br>It look strange to me. I expected FreeSwitch to at least send some <br>"empty" RTP packets to SIP endpoints as soon as call estabilishment is <br>confirmed on SIP channel. But Freeswitch doesn't do it and only sends <br>ringing indication.<br><br>I had problems in scenario, where SIP2 endpoint doesn't ring and <br>immediately answers the call. Call
was estabilished, but audio was stuck <br>in our firewall, because Freeswitch haven't initiated any RTP <br>communication yet and was probably waiting for something.<br><br>I managed to solve this problem by adding <action application="set" <br>data="instant_ringback=true"/> between my <action <br>application="ring_ready" /> and <action application="bridge" /> commands <br>in dialplan.<br><br>Now it works, but I'm feeling kind of uncertain with this solution, <br>because I don't really understand why. What was Freeswitch waiting for, <br>before I added instant_ringback? Why can't FreeSwitch prepare NAT holes <br>(as described before) as soon as SDP are interchanged?<br><br>Thanks for clarification.<br><br>Regards<br><br>-- <br>Michal Zubac<br>ComGate Interactive s.r.o.<br><br>Prague Marina Office Center<br>Jankovcova 1596/14a<br>17000 Praha 7, Czech
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