[SR-Users] Call drops after 1 minute

[Henning Westerholt]

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 <cbw at itf-as.no>
Sent: Wednesday, December 7, 2022 8:14 AM
To: Henning Westerholt <hw at gilawa.com>
Cc: Kamailio (SER) - Users Mailing List <sr-users at 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<http://213.52.37.107:50336> -> 10.1.2.10:5060<http://10.1.2.10:5060> #1
  INVITE sip:kmm at sip2.itf-as.com<mailto:sip%3Akmm at 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 at sip2.itf-as.com<mailto:sip%3Acbwlap at sip2.itf-as.com>>;tag=4183d760c26e4531a7a39f45d1
  4fb4c6..To: <sip:kmm at sip2.itf-as.com<mailto:sip%3Akmm at sip2.itf-as.com>>..Contact: <sip:cbwlap at 213.52.37.107:35270;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:cbwlap at sip2.itf-as.com<mailto:sip%3Acbwlap at sip2.itf-as.com>>;tag=4183d760c26e
  4531a7a39f45d14fb4c6..To: <sip:kmm at sip2.itf-as.com<mailto:sip%3Akmm at sip2.itf-as.com>>;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:kmm at sip2.itf-as.com<mailto:sip%3Akmm at 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 at sip2.itf-as.com<mailto:sip%3Acbwlap at sip2.itf-as.com>>;tag=4183d760c26e4531a7a39f45d14fb
  4c6..To: <sip:kmm at sip2.itf-as.com<mailto:sip%3Akmm at sip2.itf-as.com>>;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 <hw at gilawa.com<mailto:hw at 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 at lists.kamailio.org<mailto:sr-users-bounces at lists.kamailio.org>> On Behalf Of Christian B Wiik
Sent: Wednesday, December 7, 2022 7:43 AM
To: sr-users at lists.kamailio.org<mailto:sr-users at 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<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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