NAT problems, FS keep sending ACK to local IP Address
Hi, I have “30 second-then-hangup” problem. I have traced the problem, and turn out that the server keeps sending ACK to SIP Client local IP Address (it is behind the NAT). Is it a misconfigure in FS? I notice that in SDP received from SIP Client the rtcp port is followed by SIP Client local IP address. Is it right or not? Here is the SIP trace result: recv 738 bytes from udp/[118.xxx.xxx.xxx]:6900 at 10:21:33.992037: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.150:57699 ;rport=57699;branch=z9hG4bK4Q9KyFrDr24gc From: "400001" <sip:400001@180.xxx.xxx.xxx>;tag=59etrFg3F0pvg To: <sip:400002@118.xxx.xxx.xxx:6900;line=b1bb925d6ab814e>;tag=189404928 Call-ID: ea145392-ddb9-1231-9c95-94de80208578 CSeq: 53085761 INVITE Contact: <sip:linphone.android@192.168.8.47:6900> Content-Type: application/sdp User-Agent: LinphoneAndroid/0.3.7 (eXosip2/3.6.0) Content-Length: 260 v=0 o=400002 1335 3400 IN IP4 118.xxx.xxx.xxx s=Talk c=IN IP4 118.xxx.xxx.xxx b=AS:380 t=0 0 m=audio 7078 RTP/AVP 98 101 a=rtpmap:98 speex/8000 a=fmtp:98 vbr=on a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7079 IN IP4 192.168.8.47 ------------------------------------------------------------------------ nta.c:3201 agent_recv_response() nta: received 200 OK for INVITE (53085761) nta.c:3264 agent_recv_response() nta: 200 OK is going to a transaction send 434 bytes to udp/[192.168.8.47]:6900 at 10:21:33.992228: ------------------------------------------------------------------------ ACK sip:linphone.android@192.168.8.47:6900 SIP/2.0 Via: SIP/2.0/UDP 180.xxx.xxx.xxx:57699;rport;branch=z9hG4bK502c0a9gNBU3Q Max-Forwards: 70 From: "400001" <sip:400001@180.xxx.xxx.xxx>;tag=59etrFg3F0pvg To: <sip:400002@118.xxx.xxx.xxx:6900;line=b1bb925d6ab814e>;tag=189404928 Call-ID: ea145392-ddb9-1231-9c95-94de80208578 CSeq: 53085761 ACK Contact: <sip:mod_sofia@180.xxx.xxx.xxx:57699> Content-Length: 0 ------------------------------------------------------------------------ nta.c:8169 outgoing_send() nta: resent ACK (53085761) to UDP/ 192.168.8.47:6900 Thanks, Regards Donny
On 12/12/13 11:48, Donny Hardyanto wrote:
Contact: <sip:linphone.android@192.168.8.47:6900 <http://sip:linphone.android@192.168.8.47:6900>> Seems like your Linphone submits it's local IP address. Can you enable STUN?
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
Even when I enabled stun, the problem is still persist. The FS is oke about 1 month ago. I can make call more that 30 second and it was very stable. And the client is not changed. I was upgrade the FS from stable git couple weeks ago, and then outbound gateway was blocked by third party operator, we did not use the system in 1-2 week. When the gateway was on again, we start using the system again, all the call is like this, even call between extension that has nothing to do with the gateway. I am trying to fix it by upgrading almost every day. But the problem is still persist. I am at loss. Any suggestion is very appreciated. Thanks, Donny On Thu, Dec 12, 2013 at 10:19 PM, Peter Steinbach <lists@telefaks.de> wrote:
On 12/12/13 11:48, Donny Hardyanto wrote:
Contact: <sip:linphone.android@192.168.8.47:6900>
Seems like your Linphone submits it's local IP address. Can you enable STUN?
-- With kind regards Peter Steinbach
Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de Internet: www.telefaks.de
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Hi Donny, See this wiki page http://wiki.freeswitch.org/wiki/ACL And change this setting <param name="local-network-acl" value="localnet.auto"/> Try these values: * rfc1918.auto - RFC 1918<http://tools.ietf.org/html/rfc1918> Space. * nat.auto - RFC 1918<http://tools.ietf.org/html/rfc1918> Excluding your local lan. * localnet.auto - ACL for your local lan. <== DEFAULT Hope this helps, Robert From: Donny Hardyanto [mailto:hardyanto.donny@gmail.com] Sent: Thursday, December 12, 2013 10:45 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] NAT problems, FS keep sending ACK to local IP Address Even when I enabled stun, the problem is still persist. The FS is oke about 1 month ago. I can make call more that 30 second and it was very stable. And the client is not changed. I was upgrade the FS from stable git couple weeks ago, and then outbound gateway was blocked by third party operator, we did not use the system in 1-2 week. When the gateway was on again, we start using the system again, all the call is like this, even call between extension that has nothing to do with the gateway. I am trying to fix it by upgrading almost every day. But the problem is still persist. I am at loss. Any suggestion is very appreciated. Thanks, Donny On Thu, Dec 12, 2013 at 10:19 PM, Peter Steinbach <lists@telefaks.de<mailto:lists@telefaks.de>> wrote: On 12/12/13 11:48, Donny Hardyanto wrote: Contact: <sip:linphone.android@192.168.8.47:6900<http://sip:linphone.android@192.168.8.47:6900>> Seems like your Linphone submits it's local IP address. Can you enable STUN? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de<http://telefaks.de> Internet: www.telefaks.de<http://www.telefaks.de> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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
Finally solved with: <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> The bug is in the client, and this parameter force FS to adapt to the bug. The reason it was fine before, because I inadvertently add this parameter when trying to create surrogate SIP client in FS, I just copy paste it. When I finish experimenting, I comment the config including this very line. Putting it back solve the problem. Viva FS! Donny
participants (3)
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Donny Hardyanto
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Peter Steinbach
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Robert Hadley