Re: [SR-Users] sipp to kamailio doesn't match ACK [SOLVED] - sr-users

16 Jun 2010

On Tue, Jun 15, 2010 at 3:44 PM, JR Richardson &lt;jmr.richardson(a)gmail.com&gt; wrote:
...
  Hi All,

 I have been in the lab testing various load balancing scenarios with
 kamailio 1.5 and 3.0 and distributing calls to asterisk servers.  I am
 running into an issue that I'm sure is related to sipp not kamailio or
 asterisk.  My testing is setup like this:
 sipp><kamailio stateful (dispatcher or carrierroute)><multiple asterisk
servers

 My sipp scenario:
 sipp -sn uac 10.10.12.22 -i 10.10.14.97 -s 22265 -d 5000 -l 100 -r 1
 -trace_err -m 1 -nr

 What I'm seeing in my ngrep sip traces is the ACK coming back from
 sipp does not have the RR header or a proper ftag:

 Here is the Ok sent from kamailio 10.10.12.22 to sipp 10.10.14.97:

 U 10.10.12.22:5060 -> 10.10.14.97:5061
 SIP/2.0 200 OK.
 Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-0.
 Record-Route: <sip:10.10.12.22;lr=on;ftag=10002SIPpTag001>.
 From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001.
 To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7.
 Call-ID: 1-10002(a)10.10.14.97.
 CSeq: 1 INVITE.
 User-Agent: Asterisk PBX.
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
 Supported: replaces.
 Contact: <sip:65@10.10.14.102>.
 Content-Type: application/sdp.
 Content-Length: 184.
 .
 v=0.
 o=root 23641 23641 IN IP4 10.10.14.102.
 s=session.
 c=IN IP4 10.10.14.102.
 t=0 0.
 m=audio 24270 RTP/AVP 0.
 a=rtpmap:0 PCMU/8000.
 a=silenceSupp:off - - - -.
 a=ptime:20.
 a=sendrecv.

 Here is the reply ACK sent from sipp 10.10.14.97 to kamailio 10.10.12.22:

 U 10.10.14.97:5061 -> 10.10.12.22:5060
 ACK sip:22265@10.10.12.22:5060 SIP/2.0.
 Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-5.
 From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001.
 To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7.
 Call-ID: 1-10002(a)10.10.14.97.
 CSeq: 1 ACK.
 Contact: sip:sipp@10.10.14.97:5061.
 Max-Forwards: 70.
 Subject: Performance Test.
 Content-Length: 0.

 And the ACK goes into the to-tag portion of the kamailio route script
 but can not match the transaction so kamailio exits, the dialog drops
 out of memory and when sipp sends a BYE, kamailio replys with 404 not
 found.

 So if I'm diagnosing this correctly, sipp is not maintaining the
 Record-Route: header in the responses back to kamailio and without
 that info kamailio can not maintain transaction state and the call
 fails.  Is there any work around or possibly another SIP performance
 testing suite that will handle RR headers?

 Thanks.

 JR
 --
 JR Richardson
 Engineering for the Masses
 After reassuring myself that my diagnosis was correct and shortly
before I went blind tracing sip messages, I dug into sipp and found
the ability to create a new scenario adding the record route header
and next_url parameter.  So now when I run the testing from sipp, the
dialogs are properly processed statefully through kamailio and all
calls complete as expected.

I though I new SIP pretty well, but it's the problems I run into that
convince me there is so much more to learn.

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses