[SR-Users] Call drops after 1 minute

[Christian B Wiik]

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 <hw at 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 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 -> 10.1.2.10:5060 #1
>   INVITE sip:kmm 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>;tag=4183d760c26e4531a7a39f45d1
>   4fb4c6..To: <sip:kmm 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;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 at sip2.itf-as.com>;tag=4183d760c26e
>   4531a7a39f45d14fb4c6..To: <sip:kmm at 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 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>;tag=4183d760c26e4531a7a39f45d14fb
>   4c6..To: <sip:kmm at 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 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> *On Behalf Of *Christian
> B Wiik
> *Sent:* Wednesday, December 7, 2022 7:43 AM
> *To:* 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 -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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