23 Dec
2022
23 Dec
'22
9:40 p.m.
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 <hw(a)gilawa.com> 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 <cbw(a)itf-as.no>
Sent: Wednesday, December 7, 2022 9:13 AM
To: Henning Westerholt <hw(a)gilawa.com>
Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Call drops after 1 minute
Link to entire trace:
https://docs.google.com/document/d/1yWFJ_Cv13p5cYk-d8m5HMBSeLalkutV0cKZHjHf…
--
Regards
Christian
ons. 7. des. 2022 kl. 08:57 skrev Henning Westerholt <hw(a)gilawa.com>om>:
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(a)itf-as.no>
Sent: Wednesday, December 7, 2022 8:14 AM
To: Henning Westerholt <hw(a)gilawa.com>
Cc: Kamailio (SER) - Users Mailing List <sr-users(a)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(a)gilawa.com>om>:
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(a)lists.kamailio.org> On Behalf Of Christian B Wiik
Sent: Wednesday, December 7, 2022 7:43 AM
To: sr-users(a)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.
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James Browne