Re: [SR-Users] Call drops after 1 minute

-- Regards Christian ons. 7. des. 2022 kl. 08:57 skrev Henning Westerholt hw@gilawa.com: > 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 cbw@itf-as.no > *Sent:* Wednesday, December 7, 2022 8:14 AM > *To:* Henning Westerholt hw@gilawa.com > *Cc:* Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org > *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:kmm@sip2.itf-as.com SIP/2.0..Via: SIP/2.0/UDP > 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9 > 706413f868bdd222cadbed8..Max-Forwards: 70..From: < > sip:cbwlap@sip2.itf-as.com>;tag=4183d760c26e4531a7a39f45d1 > 4fb4c6..To: sip:kmm@sip2.itf-as.com..Contact: < > sip:cbwlap@213.52.37.107:35270;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:cbwlap@sip2.itf-as.com>;tag=4183d760c26e > 4531a7a39f45d14fb4c6..To: sip:kmm@sip2.itf-as.com >;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:kmm@sip2.itf-as.com SIP/2.0..Via: SIP/2.0/UDP > 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9706 > 413f868bdd222cadbed8..Max-Forwards: 70..From: < > sip:cbwlap@sip2.itf-as.com>;tag=4183d760c26e4531a7a39f45d14fb > 4c6..To: sip:kmm@sip2.itf-as.com;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 hw@gilawa.com: > > 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 sr-users-bounces@lists.kamailio.org *On Behalf Of *Christian > B Wiik > *Sent:* Wednesday, December 7, 2022 7:43 AM > *To:* sr-users@lists.kamailio.org > *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. > > > >