Hello, it looks that the 200 OK is not properly transmitted, as they seem to be retransmits. Maybe you should read a bit about SIP if you have not done it yet. It will help you debugging this problem (and probably others in the future). 😉
Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Christian B Wiik <[email protected]> Sent: Wednesday, December 7, 2022 9:13 AM To: Henning Westerholt <[email protected]> Cc: Kamailio (SER) - Users Mailing List <[email protected]> Subject: Re: [SR-Users] Call drops after 1 minute Link to entire trace: https://docs.google.com/document/d/1yWFJ_Cv13p5cYk-d8m5HMBSeLalkutV0cKZHjHf1QHk/edit?usp=sharing -- Regards Christian ons. 7. des. 2022 kl. 08:57 skrev Henning Westerholt <[email protected]<mailto:[email protected]>>: Hi Christian, this ACK is the reply to the 407 and not the relevant one for the dialog. Please have a look to the full SIP dialog. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Christian B Wiik <[email protected]<mailto:[email protected]>> Sent: Wednesday, December 7, 2022 8:14 AM To: Henning Westerholt <[email protected]<mailto:[email protected]>> Cc: Kamailio (SER) - Users Mailing List <[email protected]<mailto:[email protected]>> Subject: Re: [SR-Users] Call drops after 1 minute Thanks Henning. These are the first 3 packets filtering on my user. I see the ACK but I'm not able to spot the error. U 213.52.37.107:50336<http://213.52.37.107:50336> -> 10.1.2.10:5060<http://10.1.2.10:5060> #1 INVITE sip:[email protected]<mailto:sip%[email protected]> SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9 706413f868bdd222cadbed8..Max-Forwards: 70..From: <sip:[email protected]<mailto:sip%[email protected]>>;tag=4183d760c26e4531a7a39f45d1 4fb4c6..To: <sip:[email protected]<mailto:sip%[email protected]>>..Contact: <sip:[email protected]:35270;ob>..Call-ID: b3dd380f0c1d4e 0ebdd7fc223710d938..CSeq: 23860 INVITE..Route: <sip:sip2.itf-as.com<http://sip2.itf-as.com>;lr>..Allow: PRACK, INVITE, ACK, BYE, CAN CEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS..Supported: replaces, 100rel, timer, norefersu b..Session-Expires: 1800..Min-SE: 90..User-Agent: MicroSIP/3.21.3.<http://3.21.3.>.Content-Type: application/sdp..Content-Le ngth: 345....v=0..o=- 3879388988 3879388988 IN IP4 213.52.37.107..s=pjmedia..b=AS:84..t=0 0..a=X-nat:0..m= audio 35276 RTP/AVP 8 0 101..c=IN IP4 213.52.37.107..b=TIAS:64000..a=rtcp:35277 IN IP4 213.52.37.107..a=send recv..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ssrc :1053777612 cname:28d400de4b7d5918.. # U 10.1.2.10:5060<http://10.1.2.10:5060> -> 213.52.37.107:50336<http://213.52.37.107:50336> #2 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 213.52.37.107:35270;rport=50336;branch=z9hG4bKPj 398365dc9706413f868bdd222cadbed8;received=213.52.37.107..From: <sip:[email protected]<mailto:sip%[email protected]>>;tag=4183d760c26e 4531a7a39f45d14fb4c6..To: <sip:[email protected]<mailto:sip%[email protected]>>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID: b3dd380f0c1d4e0ebdd7fc223710d938..CSeq: 23860 INVITE..Proxy-Authenticate: Digest realm="sip2.itf-as.com<http://sip2.itf-as.com>", no nce="Y5A72WOQOq3afsXxs6AD2ihlmLAlgNOe"..Server: kamailio (5.6.2 (x86_64/linux))..Content-Length: 0.... # U 213.52.37.107:50336<http://213.52.37.107:50336> -> 10.1.2.10:5060<http://10.1.2.10:5060> #3 ACK sip:[email protected]<mailto:sip%[email protected]> SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9706 413f868bdd222cadbed8..Max-Forwards: 70..From: <sip:[email protected]<mailto:sip%[email protected]>>;tag=4183d760c26e4531a7a39f45d14fb 4c6..To: <sip:[email protected]<mailto:sip%[email protected]>>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID: b3dd380f0c1d4e0eb dd7fc223710d938..CSeq: 23860 ACK..Route: <sip:sip2.itf-as.com<http://sip2.itf-as.com>;lr>..Content-Length: 0.... -- Regards Christian ons. 7. des. 2022 kl. 07:51 skrev Henning Westerholt <[email protected]<mailto:[email protected]>>: Hello, as you’ve guessed, this can be a common problem related to the routing of the ACK message. Have a look e.g. with ngrep or sngrep to the SIP signalisation on the server side and check if everything is correct in the SIP messages. From: sr-users <[email protected]<mailto:[email protected]>> On Behalf Of Christian B Wiik Sent: Wednesday, December 7, 2022 7:43 AM To: [email protected]<mailto:[email protected]> Subject: [SR-Users] Call drops after 1 minute Greetings! I have a CentOS setup in AWS where all my calls are dropped after about a minute or so. I realize this typically is a NAT problem, but I can't see where my error is. Sound is fine both ways. Kamailio is set with WITH_NAT and I use rtpproxy like this: OPTIONS="-l 10.1.2.10 -s udp:127.0.0.1:7722<http://127.0.0.1:7722> -d INFO:LOG_LOCAL5 -m 35010 -M 35110 -A 54.171.168.48" (10.1.2.10 is the local IP for CentOS) Tested with MicroSIP and Linphone and tried numerous configurations. It seems the receiving client is not able to verify the call has been set up, and disconnects. MicroSIP has the status "Connecting..." until it disconnects. All tips appreciated. Will post configuration and logs if needed. Kamailio version 5.6.2 from rpm and rtpproxy 2.1.0 compiled from source.
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