7 Dec
2022
7 Dec
'22
10:57 a.m.
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(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<http://213.52.37.107:50336> ->
10.1.2.10:5060<http://10.1.2.10:5060> #1
INVITE sip:kmm@sip2.itf-as.com<mailto:sip%3Akmm@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<mailto:sip%3Acbwlap@sip2.itf-as.com>>;tag=4183d760c26e4531a7a39f45d1
4fb4c6..To:
<sip:kmm@sip2.itf-as.com<mailto:sip%3Akmm@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<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@sip2.itf-as.com<mailto:sip%3Acbwlap@sip2.itf-as.com>>;tag=4183d760c26e
4531a7a39f45d14fb4c6..To:
<sip:kmm@sip2.itf-as.com<mailto:sip%3Akmm@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@sip2.itf-as.com<mailto:sip%3Akmm@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<mailto:sip%3Acbwlap@sip2.itf-as.com>>;tag=4183d760c26e4531a7a39f45d14fb
4c6..To:
<sip:kmm@sip2.itf-as.com<mailto:sip%3Akmm@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@gilawa.com<mailto: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<mailto: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<mailto: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<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.