<div dir="ltr">Link to entire trace:<div><br><div><a href="https://docs.google.com/document/d/1yWFJ_Cv13p5cYk-d8m5HMBSeLalkutV0cKZHjHf1QHk/edit?usp=sharing">https://docs.google.com/document/d/1yWFJ_Cv13p5cYk-d8m5HMBSeLalkutV0cKZHjHf1QHk/edit?usp=sharing</a></div><div><br></div><div>-- <br clear="all"><div><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Regards</div><div>Christian</div></div></div></div><br></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">ons. 7. des. 2022 kl. 08:57 skrev Henning Westerholt <<a href="mailto:hw@gilawa.com">hw@gilawa.com</a>>:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class="msg8062202936681656049">
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<p class="MsoNormal"><span>Hi Christian,<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB">this ACK is the reply to the 407 and not the relevant one for the dialog.<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB">Please have a look to the full SIP dialog.<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB">Cheers,<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB">Henning<u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB"><u></u> <u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB">-- <u></u>
<u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB">Henning Westerholt –
</span><span><a href="https://skalatan.de/blog/" target="_blank"><span lang="EN-GB" style="color:rgb(5,99,193)">https://skalatan.de/blog/</span></a></span><span lang="EN-GB"><u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB">Kamailio services –
</span><span><a href="https://gilawa.com/" target="_blank"><span lang="EN-GB" style="color:rgb(5,99,193)">https://gilawa.com</span></a></span><span lang="EN-GB"><u></u><u></u></span></p>
<p class="MsoNormal"><span lang="EN-GB"><u></u> <u></u></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><b>From:</b> Christian B Wiik <<a href="mailto:cbw@itf-as.no" target="_blank">cbw@itf-as.no</a>>
<br>
<b>Sent:</b> Wednesday, December 7, 2022 8:14 AM<br>
<b>To:</b> Henning Westerholt <<a href="mailto:hw@gilawa.com" target="_blank">hw@gilawa.com</a>><br>
<b>Cc:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a>><br>
<b>Subject:</b> Re: [SR-Users] Call drops after 1 minute<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><u></u> <u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt">Thanks Henning.<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><u></u> <u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt">These are the first 3 packets filtering on my user. I see the ACK but I'm not able to spot the error.<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><u></u> <u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt">U <a href="http://213.52.37.107:50336" target="_blank">
213.52.37.107:50336</a> -> <a href="http://10.1.2.10:5060" target="_blank">10.1.2.10:5060</a> #1<br>
INVITE <a href="mailto:sip%3Akmm@sip2.itf-as.com" target="_blank">sip:kmm@sip2.itf-as.com</a> SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9<br>
706413f868bdd222cadbed8..Max-Forwards: 70..From: <<a href="mailto:sip%3Acbwlap@sip2.itf-as.com" target="_blank">sip:cbwlap@sip2.itf-as.com</a>>;tag=4183d760c26e4531a7a39f45d1<br>
4fb4c6..To: <<a href="mailto:sip%3Akmm@sip2.itf-as.com" target="_blank">sip:kmm@sip2.itf-as.com</a>>..Contact: <<a>sip:cbwlap@213.52.37.107:35270;ob</a>>..Call-ID: b3dd380f0c1d4e<br>
0ebdd7fc223710d938..CSeq: 23860 INVITE..Route: <sip:<a href="http://sip2.itf-as.com" target="_blank">sip2.itf-as.com</a>;lr>..Allow: PRACK, INVITE, ACK, BYE, CAN<br>
CEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS..Supported: replaces, 100rel, timer, norefersu<br>
b..Session-Expires: 1800..Min-SE: 90..User-Agent: MicroSIP/<a href="http://3.21.3." target="_blank">3.21.3.</a>.Content-Type: application/sdp..Content-Le<br>
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=<br>
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<br>
recv..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ssrc<br>
:1053777612 cname:28d400de4b7d5918..<br>
#<br>
U <a href="http://10.1.2.10:5060" target="_blank">10.1.2.10:5060</a> -> <a href="http://213.52.37.107:50336" target="_blank">
213.52.37.107:50336</a> #2<br>
SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 213.52.37.107:35270;rport=50336;branch=z9hG4bKPj<br>
398365dc9706413f868bdd222cadbed8;received=213.52.37.107..From: <<a href="mailto:sip%3Acbwlap@sip2.itf-as.com" target="_blank">sip:cbwlap@sip2.itf-as.com</a>>;tag=4183d760c26e<br>
4531a7a39f45d14fb4c6..To: <<a href="mailto:sip%3Akmm@sip2.itf-as.com" target="_blank">sip:kmm@sip2.itf-as.com</a>>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID:<br>
b3dd380f0c1d4e0ebdd7fc223710d938..CSeq: 23860 INVITE..Proxy-Authenticate: Digest realm="<a href="http://sip2.itf-as.com" target="_blank">sip2.itf-as.com</a>", no<br>
nce="Y5A72WOQOq3afsXxs6AD2ihlmLAlgNOe"..Server: kamailio (5.6.2 (x86_64/linux))..Content-Length: 0....<br>
#<br>
U <a href="http://213.52.37.107:50336" target="_blank">213.52.37.107:50336</a> -> <a href="http://10.1.2.10:5060" target="_blank">
10.1.2.10:5060</a> #3<br>
ACK <a href="mailto:sip%3Akmm@sip2.itf-as.com" target="_blank">sip:kmm@sip2.itf-as.com</a> SIP/2.0..Via: SIP/2.0/UDP 213.52.37.107:35270;rport;branch=z9hG4bKPj398365dc9706<br>
413f868bdd222cadbed8..Max-Forwards: 70..From: <<a href="mailto:sip%3Acbwlap@sip2.itf-as.com" target="_blank">sip:cbwlap@sip2.itf-as.com</a>>;tag=4183d760c26e4531a7a39f45d14fb<br>
4c6..To: <<a href="mailto:sip%3Akmm@sip2.itf-as.com" target="_blank">sip:kmm@sip2.itf-as.com</a>>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.f003010a..Call-ID: b3dd380f0c1d4e0eb<br>
dd7fc223710d938..CSeq: 23860 ACK..Route: <sip:<a href="http://sip2.itf-as.com" target="_blank">sip2.itf-as.com</a>;lr>..Content-Length: 0....<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><u></u> <u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt">-- <br clear="all">
<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt">Regards<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt">Christian<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><u></u> <u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><u></u> <u></u></p>
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<p class="MsoNormal" style="margin-left:35.4pt">ons. 7. des. 2022 kl. 07:51 skrev Henning Westerholt <<a href="mailto:hw@gilawa.com" target="_blank">hw@gilawa.com</a>>:<u></u><u></u></p>
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<blockquote style="border-top:none;border-right:none;border-bottom:none;border-left:1pt solid rgb(204,204,204);padding:0cm 0cm 0cm 6pt;margin-left:4.8pt;margin-right:0cm">
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<p class="MsoNormal" style="margin-left:35.4pt">
Hello,<u></u><u></u></p>
<p class="MsoNormal" style="margin-left:35.4pt">
<u></u><u></u></p>
<p class="MsoNormal" style="margin-left:35.4pt">
<span lang="EN-GB">as you’ve guessed, this can be a common problem related to the routing of the ACK message.</span><u></u><u></u></p>
<p class="MsoNormal" style="margin-left:35.4pt">
<span lang="EN-GB"> </span><u></u><u></u></p>
<p class="MsoNormal" style="margin-left:35.4pt">
<span lang="EN-GB">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.</span><u></u><u></u></p>
<p class="MsoNormal" style="margin-left:35.4pt">
<span lang="EN-GB"> </span><u></u><u></u></p>
<p class="MsoNormal" style="margin-left:35.4pt">
<u></u> <u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
<b>From:</b> sr-users <<a href="mailto:sr-users-bounces@lists.kamailio.org" target="_blank">sr-users-bounces@lists.kamailio.org</a>>
<b>On Behalf Of </b>Christian B Wiik<br>
<b>Sent:</b> Wednesday, December 7, 2022 7:43 AM<br>
<b>To:</b> <a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a><br>
<b>Subject:</b> [SR-Users] Call drops after 1 minute<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
Greetings!<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
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.<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
Sound is fine both ways.<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
Kamailio is set with WITH_NAT and I use rtpproxy like this:<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
OPTIONS="-l 10.1.2.10 -s udp:<a href="http://127.0.0.1:7722" target="_blank">127.0.0.1:7722</a> -d INFO:LOG_LOCAL5 -m 35010 -M 35110 -A 54.171.168.48"<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
(10.1.2.10 is the local IP for CentOS)<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
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.<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
All tips appreciated. Will post configuration and logs if needed.<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
Kamailio version 5.6.2 from rpm and rtpproxy 2.1.0 compiled from source.<u></u><u></u></p>
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<p class="MsoNormal" style="margin-left:70.8pt">
<u></u> <u></u></p>
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