Re: [SR-Users] Call drops after 1 minute

7 Dec 2022

      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@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 *On Behalf Of *Christian
> B Wiik
> *Sent:* Wednesday, December 7, 2022 7:43 AM
> *To:* 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 -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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Re: [SR-Users] Call drops after 1 minute