Hi Antoine,

As previously mentioned, the new "DontPreRegisterButUseSippIP" option allows 
the pre-registration phase to be skipped.
This means that the users' SIP address directly contains the "@IP:port" of the 
SIPp instance, instead of "@open-ims.test" which would require a DNS and an HSS 
in order to be resolved.
This also allows to consider users as already registered when the test starts, 
so they can immediately be part of a call.

Here is a quick reminder about how IMS Bench SIPp default configuration works:
- at the beginning, all users are assigned to pool 0
- during the pre-registration phase, the ims_reg scenario transiently moves 
users from pool 0 to pool 1, and then to pool 2 once users have been registered
- during the real test phase, the ims_uac and ims_uas scenarios pick users from 
pool 2, temporarily move them to pool 3 during the call, and return them to 
pool 2 at the end of the call

Using the "DontPreRegisterButUseSippIP" option, the SailFin-dedicated 
configuration works as follows:
- at the beginning, all users are assigned to pool 0
- during the real test phase, the sailfin_uac and sailfin_uas scenarios pick 
users from pool 0, temporarily move them to pool 2 during the call, and return 
them to pool 0 at the end of the call

So, the pick_user() line is not strange, because users must be selected for the 
call from a pool where they are considered as "registered", or at least as 
"ready to be part of a call".
I am a bit more concerned about the "User pool id[0] is empty -> Quitting!." 
line...

Basically, this error means that there are no more users available in pool 0, 
which suggests that all users are currently part of a call and have been 
temporarily moved to pool 2.
There might be several reasons for that problem:
- either you did not provision enough users (using the 
"TotalProvisionedSubscribers" option), with respect to the load (SAPS) and the 
average duration of calls (HoldTime)
- or users are never returned from pool 2 to pool 0 at the end of the call, for 
example because the SUT does not properly forward the BYE message...

Since your problem occurs at Call-Id 10986, I think it's the first reason here, 
and I guess you only provisioned about 20000 users. You need many more of them!
If you can't figure out how to fix this, then please send me the ims_bench.xml 
file that was generated by the ims_bench.pl script into the newly-created 
ims_bench_XXX/ directory.

Regards,
Patrice

-----Original Message-----
From: Antoine Roly [mailto:antoine.r...@gmail.com] 
Sent: Wednesday, May 12, 2010 11:33 AM
To: Buriez, Patrice
Cc: sipp-users@lists.sourceforge.net
Subject: RE: [Sipp-users] sipp ims bench

Hi Patrice,

First of all thank you for your response.

I tried the solution you told me, it almost works... 
In fact, the test starts correctly, I see the SIP messages sent and
received in the SIPp instance running on the Test System. After some
time, SIPp stops with the message 

"sipp.cpp: EXIT_TEST_RES_UNKNOWN quitting=1".

In the error log file on the TS, i can see these lines

sipp: The following events occured:
Created CConsole 0x9c3b488.
SUT h=10 l=4 SO_REUSEADDR=1.
SUT h=10 l=4 TCP_NODELAY=0.
SUT h=10 l=8 SO_LINGER=(1,1).
SUT h=10 l=4 SO_SNDBUF=262142.
SUT h=10 l=4 SO_RCVBUF=262142.
SUT h=11 l=4 SO_REUSEADDR=1.
SUT h=11 l=4 TCP_NODELAY=0.
SUT h=11 l=8 SO_LINGER=(1,1).
SUT h=11 l=4 SO_SNDBUF=262142.
SUT h=11 l=4 SO_RCVBUF=262142.
Set TSID: slot:0  TS1 (1).
accept return 18 [IP4: 192.168.4.100:47710].
~ASSIGNID=0xffffffff - 0x1.
~ASSIGNID=0x1 - 0xffffffff.
Set TSID: slot:1  TS1 (0).
User pool id[0] is empty -> Quitting!.
Quitting!.
Call '10986-1...@192.168.4.100' - pick_user() returned NULL.
Call '10986-1...@192.168.4.100' - Action 'move_user' without a user
assigned!.
!! ERROR !! There should have been calls in SYNC !!.
C:'10985-1...@192.168.4.100', T/O sailfin_uac:2 without ontimeout.
C:'10985-1...@192.168.4.100' [0xc0615d0/10985] Aborting! sailfin_uac:2.
2010-05-12 10:38:50.526: CSutSocket::HandleEvent POLLIN: could not find
a call for Call-ID "10985-1...@192.168.4.100".
CSutSocket: Enter congestion h=11 s=502 ret=166 o=336.
CSutSocket::HandleEvent POLLOUT h=11 s=565831 ret=89776.
CSutSocket::HandleEvent POLLOUT h=11 s=552917 ret=82536.
CSutSocket::HandleEvent POLLOUT h=11 s=526629 ret=89776.
CSutSocket::HandleEvent POLLOUT h=11 s=439378 ret=88328.
CSutSocket::HandleEvent POLLOUT h=11 s=351050 ret=86880.
CSutSocket::HandleEvent POLLOUT h=11 s=264170 ret=88328.
CSutSocket::HandleEvent POLLOUT h=11 s=177353 ret=85432.
CSutSocket::HandleEvent POLLOUT h=11 s=91921 ret=86880.
CSutSocket::HandleEvent POLLOUT h=11 s=5041 ret=5041.
CSutSocket: Leaving congestion... h=11 ret=5041.
sipp.cpp: EXIT_TEST_RES_UNKNOWN quitting=1.
final cleanup.

On the manager side, all I see is TS1 is deregistered and the test
stops.

Did I made a mistake, or forgot something? 

The line with the pick_user() seems strange to me, indeed there's no
preregistration in this case, so obviously if the system try to pick a
registered user it will fail... 

Do you have any idea where the problem could come from? 

Thanks in advance

A.



Le lundi 10 mai 2010 à 18:22 +0100, Buriez, Patrice a écrit :
> Hi Antoine,
> 
> IMS Bench SIPp now works with SailFin, since SVN revision 587, so please make 
> sure that you "svn co" a "recent" version.
> This was tested using SailFin v1-b60g, jdk-6u14, and a sample Invite servlet, 
> which were downloaded from:
> https://sailfin.dev.java.net/downloads/downloads.html
> http://java.sun.com/javase/downloads/index.jsp (now listing jdk-6u20), and
> https://sailfin.dev.java.net/source/browse/sailfin/sailfin-tests/community/systemtest/
> 
> On the SUT, install JDK and SailFin as documented, then unzip 
> InviteServlet.zip and install the servlet as follows:
> - export PATH, JAVA_HOME and ANT_HOME as appropriate
> - start domain1 sample domain:
>   cd /your/path/to/sailfin/
>   ./bin/asadmin start-domain domain1
> - deploy Invite servlet:
>   ./bin/asadmin deploy 
> /your/path/to/the/unzipped/InviteServlet/dist/InviteServlet.war
> - stop domain1 sample domain:
>   ./bin/asadmin stop-domain domain1
> - you might want to adjust the SailFin SIP session timeout to 15, in order to 
> prevent the SUT from dropping calls, in both:
>   domains/domain1/applications/j2ee-modules/InviteServlet/WEB-INF/sip.xml and
>   domains/domain1/generated/xml/j2ee-modules/InviteServlet/WEB-INF/sip.xml
> - then restart domain1 sample domain:
>   ./bin/asadmin start-domain domain1
> 
> Install IMS Bench SIPp on the test systems as per the instructions provided 
> at http://sipp.sourceforge.net/ims_bench/reference.html#Installation
> Then, instead of using the ims_bench.pl script with its default IMS UDP 
> configuration, use it as follows:
>   cd /your/path/to/ims/bench/sipp/
>   ./scripts/ims_bench.pl scripts/sailfin.xmt
> Navigate the menus to define the IP addresses of the SUT and TS instances.
> Please note that the Traffic Set relies on the provided sailfin_uac (and 
> implicit sailfin_uas) scenarios.
> You will also notice that the new options "TransportTCP" and 
> "DontPreRegisterButUseSippIP" are set to "1". TCP is required for SailFin, 
> while skipping registration allows benchmarking a SUT that does not implement 
> the HSS (such as a simple proxy).
> Quit and "generate the benchmark run files", then run IMS Bench SIPp as 
> usual. It should work!
> 
> Regards,
> Patrice
> 
> -----Original Message-----
> From: Antoine Roly [mailto:antoine.r...@gmail.com] 
> Sent: Monday, May 10, 2010 3:55 PM
> To: sipp-users@lists.sourceforge.net
> Subject: [Sipp-users] sipp ims bench
> 
> Hi all,
> 
> I'd like to test a small SIP server running on Sailfin, and made some
> overload test. Is it possible with SIPp IMS Bench? I have read the doc
> and saw that "IMS Bench SIPp should work in the situation just the same
> way as it works with SUTs that simply proxy the cal" so I suppose it is
> possible. Am I right?
> 
> I have just installed the software and run ./scripts/ims_bench.pl. I was
> expecting to have two SIPp instances configured, while the  SIP proxy
> will be between these instances but both of them act like SIP UAC.
> Should I write myself the scenario xml file and run SIPp on the UAS
> side, while the two TS will generate trafic? Or is it possible to run
> benchmarks with specific scenario files, in this case SIPp will generate
> trafic with Poisson distribution but using my files?
> 
> And is the use of OpenIMSCore mandatory? Or could I use SIPp with other
> SIP softwares?
> 
> I'm a bit confused, did I made a mistake or not understood something
> about the soft?
> 
> Thanks in advance
> 
> A.
> 
> 
> 
> 
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