Hi Christian I read your trace. There's no ACK after the SIP-200 response. It's cumbersome to read the trace because the client and server are at the same IP address, but I did it anyway. There's no ACK after the SIP-200 response, so it looks like the client (213.52.37.107) either doesn't sent it, or sends it directly to the server (213.52.37.107, the same IP address) without properly using the RR header field in the SIP-200 response.
If you can set up a similar test and have a different host set up for the client and server (instead of 213.52.37.107 for both), and if you then trace the traffic again at all hosts, then you may find the rogue ACK request or be able to prove that the client never sent it (which might prove a broken client). When you use the client and server at the same IP address, it's not always straightforward to trace any traffic that might go directly from client to server, because it won't leave any network interface. Also I find strange that your call ends after "about a minute or so". I expect that they should end after about 31.5 seconds. It's almost certainly a SIP problem, and not any RTP/rtpproxy problem, so I recommend that you focus on the SIP. (I wrote this message before noticing that the thread is two weeks old.) James On Wed, 7 Dec 2022 at 08:41, Henning Westerholt <[email protected]> wrote: > > 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 > > > > 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]>: > > 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 > > > > From: Christian B Wiik <[email protected]> > Sent: Wednesday, December 7, 2022 8:14 AM > To: Henning Westerholt <[email protected]> > Cc: Kamailio (SER) - Users Mailing List <[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 -> 10.1.2.10:5060 #1 > INVITE 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]>;tag=4183d760c26e4531a7a39f45d1 > 4fb4c6..To: <sip:[email protected]>..Contact: > <sip:[email protected]:35270;ob>..Call-ID: b3dd380f0c1d4e > 0ebdd7fc223710d938..CSeq: 23860 INVITE..Route: > <sip: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..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 -> 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]>;tag=4183d760c26e > 4531a7a39f45d14fb4c6..To: > <sip:[email protected]>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID: > b3dd380f0c1d4e0ebdd7fc223710d938..CSeq: 23860 INVITE..Proxy-Authenticate: > Digest realm="sip2.itf-as.com", no > nce="Y5A72WOQOq3afsXxs6AD2ihlmLAlgNOe"..Server: kamailio (5.6.2 > (x86_64/linux))..Content-Length: 0.... > # > U 213.52.37.107:50336 -> 10.1.2.10:5060 #3 > ACK 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]>;tag=4183d760c26e4531a7a39f45d14fb > 4c6..To: > <sip:[email protected]>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID: > b3dd380f0c1d4e0eb > dd7fc223710d938..CSeq: 23860 ACK..Route: > <sip:sip2.itf-as.com;lr>..Content-Length: 0.... > > > > -- > > Regards > > Christian > > > > > > ons. 7. des. 2022 kl. 07:51 skrev Henning Westerholt <[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]> On Behalf Of Christian B > Wiik > Sent: Wednesday, December 7, 2022 7:43 AM > To: [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 -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. > > > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > [email protected] > Important: keep the mailing list in the recipients, do not reply only to the > sender! > Edit mailing list options or unsubscribe: > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to [email protected] Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
