On Tue, Jun 15, 2010 at 3:44 PM, JR Richardson <jmr.richard...@gmail.com> 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:s...@10.10.14.97:5061>;tag=10002SIPpTag001. > To: sut <sip:22...@10.10.12.22:5060>;tag=as18ae7aa7. > Call-ID: 1-10...@10.10.14.97. > CSeq: 1 INVITE. > User-Agent: Asterisk PBX. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Contact: <sip:6...@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:22...@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:s...@10.10.14.97:5061>;tag=10002SIPpTag001. > To: sut <sip:22...@10.10.12.22:5060>;tag=as18ae7aa7. > Call-ID: 1-10...@10.10.14.97. > CSeq: 1 ACK. > Contact: sip:s...@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 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
