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