The default SIPp CSV separator is not in fact a comma, but rather a
semi-colon. This means that if you try to load the file directly in Excel
you don't get an import wizard; and the data is unusable. Does anyone
have objections to changing the default separator to a comma after the 3.0
. Please let us know which latest version
of
SIPp we can use for Sun_OS-5.10.
Regards,
Pradeep
-Original Message-
From: Charles P Wright [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 26, 2007 8:02 PM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED];
[EMAIL PROTECTED];
SIPp
deleted by Charles P
Wright/Watson/IBM] [attachment subscribe_notify.xml deleted by
Charles P Wright/Watson/IBM]
-
This SF.net email is sponsored by: Microsoft
Defy all challenges. Microsoft(R) Visual Studio 2005.
http
There are two things that come to mind here, one that you can probably get
some reasonable information now; but will miss things like unexpected
messages. One thing that would make this nicer is if there were key words
like [rtd number=2], etc. The best way may be to use the error or short
[EMAIL PROTECTED] wrote on 09/27/2007 08:45:12 AM:
Registration works fine. And the INVITE package will send with the right
Port to the UAS.
But Sipp discard this massage because the CallID dosn't match
(reference.html#Unexpected+messages)
but i Need to Reset the callID because Register
You can use the -i option.
Charles
[EMAIL PROTECTED] wrote on 09/27/2007 05:08:52 PM:
Is there a way with sipp to specify the interface from which the
call is generated? I have a dual interface machine on different
networks and believe the attempted call is heading out the wrong
. Dexterity has taken
responsible protection to prevent this risk and accepts no
liability for any damage caused by any virus transmitted by
this mail.[attachment winmail.dat deleted by Charles P
Wright/Watson/IBM
run multiple instances of sip client acting as asterisk
extensions from a single PC?
Regards
Karthik.Ajavascript:SetCmd(cmdSend);
-Original Message-
From: Charles P Wright [mailto:[EMAIL PROTECTED]
Sent: Wed 9/26/2007 6:38 PM
To: Arumugam, Karthik [Dexterity]
Cc: sipp-users
Make sure that you can do nslookup rhc or ping -c 1 rhc.
Charles
kovendan [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
09/21/2007 03:57 AM
To
Sipp-users@lists.sourceforge.net
cc
Subject
[Sipp-users] required clarification
Hi all,
Can anyone help me how to rectify the error Can't get
Also, you may try export TERM=vt100.
Charles
[EMAIL PROTECTED] wrote on 09/20/2007 05:50:04 AM:
You forgot the ?sf option before your scenario name. Make also sure
you set your DISPLAY properly.
Olivier Boulkroune
De : [EMAIL PROTECTED] [mailto:sipp-users-
[EMAIL PROTECTED] De la
Marc,
The keyword should be [routes] without the colon.
Charles
[EMAIL PROTECTED] wrote on 09/13/2007 03:38:31 PM:
Hi,
I saw a email from Oliver Boulkroune (Re: [Sipp-users] [last_Via:}
is dropping characters) on 6-25-07 regarding the fix for last_via
dropping characters, so I
documentation to check the correct syntax.
Cheers,
Marc
On 9/13/07, Charles P Wright [EMAIL PROTECTED] wrote:
Marc,
The keyword should be [routes] without the colon.
Charles
[EMAIL PROTECTED] wrote on 09/13/2007 03:38:31
PM:
Hi,
I saw a email from Oliver Boulkroune (Re: [Sipp-users
You could try using a heap profiling tool like massif to see if you can
get any ideas.
Charles
[EMAIL PROTECTED] wrote on 09/11/2007 07:25:52 PM:
Hi,
Now I am using SIPP TLS connections to test one sip server, but what
surprised me is that the memory usage per TLS connection in SIP
Andreas,
It should be relatively straight forward to adapt the standard UAC/UAS
scenarios to do this, as long as you want only the UAS to send the bye (if
you want to make it 50/50 or something more complicated, you'll need to
use 3pcc or some other synchronization mechanism). Just copy and
You should try - instead of ~ for the ranges in your regular
expressions (e.g., [0-9a-z]).
Charles
[EMAIL PROTECTED] wrote on 08/30/2007 01:56:52 PM:
hi,
i have a problem, in the scenario,i want to extract this parameter of
the
sip message.
it include this header
Contact:sip:[EMAIL
This is not easy to accomplish with a standard script, but if you make the
decision ahead of time about who will hang up you can probably communicate
it from the UAC to the UAS using custom headers or third party call
control.
Charles
[EMAIL PROTECTED] wrote on 08/23/2007 10:48:42 AM:
Hi,
You must use a recent subversion trunk version (at least r275).
Also, the more recent versions will complain about bad actions.
Charles
K L [EMAIL PROTECTED] wrote on 08/10/2007 05:23:07 AM:
I'm using the 2.0.1 release.
On 8/9/07, Charles P Wright [EMAIL PROTECTED] wrote:
The assignstr
The assignstr action was not defined until after the 2.0 release.
Charles
[EMAIL PROTECTED] wrote on 08/09/2007 12:35:40 PM:
I have the same issue when trying to expand (use/apply) stored
variable, results comes up blank in the outgoing BYE To field. The
assignment is not being made. I'm
Jaime,
Also, what version are you using? The latest trunk versions should
support an arbitrary number of variables.
Charles
Boulkroune, Olivier (Non-HP:Atos Origin) [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
08/06/2007 04:45 AM
To
Jaime Cabrera [EMAIL PROTECTED],
You can not use a variables for most of the XML parameters, including
play_pcap_audio.
To make this work you would need to modify SIPp itself as described in the
message you cited.
Charles
[EMAIL PROTECTED] wrote on 07/31/2007 09:35:46 AM:
Hi ,
I've been watching the thread on this
Andrew,
In the intial action where you are assigning a value to 3, you should
verify that [peer_tag_param] itself produces output with a logging action.
Charles
[EMAIL PROTECTED] wrote on 07/31/2007 10:37:34 AM:
Hello members,
This is my very first post in this forum. 1 month experience in
I have successfully generated SIMPLE traffic with SIPp, you should be able
to easily adapt the default UAC and UAS scenarios by changing the BYE
message to a MESSAGE message and removing the INVITE transaction.
Charles
--
Dr. Charles P. Wright
Research Staff Member
Network Server Systems
much.
Cheers!
Zou
Jia
-Original Message-
From: Charles P Wright [EMAIL PROTECTED]
Date: Fri, 13 Jul 2007 09:23:54 -0400
To: Boulkroune, Olivier (Non-HP:Atos Origin)
[EMAIL PROTECTED]
Cc: sipp-users
=3 distribution=uniform min=0 max=1 /
log message=[call_id]: Uniform (0, 1): [$3] * [field0] /
multiply assign_to=3 variable=2 /
/action
/nop
pause variable=3 /
nop
action
log message=[call_id]: Paused until [clock_tick] /
/action
/nop
Charles
--
Dr. Charles P
Don,
This isn't supported right now, but I'll whip something up that allows you
to get it done.
Charles
[EMAIL PROTECTED] wrote on 07/05/2007 12:32:44 PM:
I'd like to program a pause that has a variable range. My problem is
that I
can't figure out how to get the variable into the pause
Olivier and Una,
The variable manipulation commands only support double values, so you can
not assign a string to it. However, based on a few emails in the last
week or two, I do think that introducing a string type (which should
probably be mostly interchangeable with the regexp type) would
Marek,
I would try inserting an optional receive, and some goto labels.
Charles
[EMAIL PROTECTED] wrote on 06/20/2007 09:34:46 AM:
Hello,
I found the option '-aa' which helps me with the INFO issue.
However, I have a similar problem with arbitrary number of periodic
re-INVITEs (they
min) there is a re-
INVITE every 2min30. I could just wait for an INVITE 12 times and
deal with each of them, but 2min30 is only the default case, in
other words it can be set to anything at all...
Thanks
Marek
De : Charles P Wright [mailto:[EMAIL PROTECTED]
Envoyé : mercredi 20
, Taiwan
胡晉華 Leo Hu
E-mail: [EMAIL PROTECTED]
Tel: +886-2-26598088 ext 6202
Charles P Wright [EMAIL PROTECTED]
Charles P Wright [EMAIL PROTECTED]
2007/06/15 下午 12:14
To
[EMAIL PROTECTED]
cc
sipp-users@lists.sourceforge.net,
[EMAIL PROTECTED]
Subject
Re: [Sipp-users
Leo,
I am glad to see that this feature would be useful for you.
1. I add $4 by 3, but in log it is still 0.
The code doesn't handle adding values to regular expressions (strings).
There should be some better error handling, or possibly even automatic
type casting added.
2. I assign $5 to
Andreas,
Do you have a label at the end of your scenario that you jump to after
your standard call setup? Something like:
recv request=INVITE crlf=true next=1 chance=0.01
/recv
send 100 /
send 180 /
send 200 next=2 /
label id=1 /
send 500 /
label id=2 /
Assuming this isn't it if you post your
Try \x5B The \x followed by two hex digits is translated into a
literal byte value, 5B corresponding to '['.
Charles
[EMAIL PROTECTED] wrote on 06/08/2007 05:09:58 AM:
folks,
I want to put the following in to a SIPP script
User-Agent: IP Phone [0.1.70]
SIPp obviously does not like
and very useful tool. Thanks
to all contributors J
Eugen
-Original Message-
From: Charles P Wright [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 30, 2007 2:57 PM
To: Peter Higginson
Cc: 'Eugen'; sipp-users@lists.sourceforge.net; sipp-users-
[EMAIL PROTECTED]
Subject: Re
Regarding #1, you should be able to even put your test on a nop, if you
can't then that is probably a feature/misdesign that needs to be
added/fixed.
Charles
[EMAIL PROTECTED] wrote on 05/30/2007 01:36:02 PM:
1) Not directly, however there is nothing to stop you putting
a test on a
Ashok,
You can use two different call-ids, but they need to be of the form
prefix1///[call_id] and prefix2///[call_id].
Charles
[EMAIL PROTECTED] wrote on 05/03/2007 08:57:28 AM:
Hi all,
Can we send 2 INVITE with different Call-ID from the same sipp
script(Basically can one sipp
only
once for
each function it appears in.)
make[1]: *** [actions.o] Error 1
looks like you're using a unit test framework ...
--Enrico
Charles P Wright wrote:
Enrico,
I apologize for the compile error. I am not yet used to SVN and
forgot to add these two files
checking?
--Enrico
Charles P Wright wrote:
Enrico,
No unit testing, but interestingly my STL headers managed to pull in
assert.h without me doing it so it compiled on my RHEL4 derived
distribution. New fix checked in.
Thanks for trying this out and having the fortitude to put up
of the second line in the sip trace)
Do you know why sipp is using the second line instead of the first one,
although 'sequential' is used?
I hope you understand my problem ...
--Enrico
Charles P Wright wrote:
Enrico,
This actually has me very confused, the file is opened exactly
Tarek,
I believe it was the short form header patch that I posted that broke
this. Another user had the same issue and I sent them this patch, but
never got any feedback. Does this fix the issue for you?
Charles
[EMAIL PROTECTED] wrote on 05/01/2007 03:38:56 PM:
Hi, I've recently
Alice's message will have a different Call-ID than Bob's existing call.
Therefore the message will not be identified as part of the conversation.
You can use three or fou r SIPp instances for this to work.
1. Bob's Registration
2. Bob's Message Receive
3. Alice (or Alice's Registration and Send)
Ashok,
The REGISTER and INVITE will have different Call-IDs. To make this work
you need to use optional messages, so that either a register or an invite
can start a new scenario.
Charles
[EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
04/26/2007 08:31 AM
To
sipp-users@lists.sourceforge.net
,
Joseph
2007-04-17 15:11 oboulkroune
* call.cpp, call.hpp: Fix: updated support of short
header forms - provided by Charles P. Wright from IBM
Research
Protocol : SIP-2.0
SIP/2.0 180 RingingCRLF
v: SIP/2.0/UDP
166.35.250.121:5060;branch
that it could hurt. :)
--
Dr. Charles P. Wright
Research Staff Member
Network Server Systems Software
IBM T.J. Watson Research Center-
This SF.net email is sponsored by DB2 Express
Download DB2 Express C - the FREE version of DB2 express
or branches and some release
engineering is going to be required.
Charles
--
Dr. Charles P. Wright
Research Staff Member
Network Server Systems Software
IBM T.J. Watson Research Center-
This SF.net email is sponsored by DB2 Express
both need to deal
with congestion, partial messages, and multiple messages).
I look forward to working with you towards a common code base,
Charles
--
Dr. Charles P. Wright
Research Staff Member
Network Server Systems Software
IBM T.J. Watson Research Center
Prem,
If your server generates a 100 trying response, the scenario will not
work; because you have no optional 100 command like:
recv response=100 optional=true /
You should insert this before the 401 receive command.
Also, what do you mean by the statistics file counter? The screen log?
We have been able to send MESSAGE messages, as far as I recall, there is
nothing special required.
Charles
[EMAIL PROTECTED] wrote on 03/28/2007 11:52:43 AM:
Hello everyone.
Is there any special requirement to be able to send SIP packets with
the MESSAGE
method?
I can't have it
Hyukgeose,
You should also be aware that on the UAS, there is a 4 second pause to
handle retransmissions, which artificially increases the call length.
Charles
[EMAIL PROTECTED] wrote on 03/23/2007 04:53:28 AM:
Hello Hyukgeose,
cps: stands for calls per second - number of new scenarios
Carl,
In general, I believe that SIPp is mostly CPU limited. You could try
reducing the frequency of the clock ticks, to reduce CPU utilization, but
that may cause other problems (e.g., increased burstiness).
We have found that to generate high loads, you will need to use more than
one
Juan,
The values are calculated by summing the response times and call lengths
in the M_counters array. To compute the average two values are needed,
the sum of the response time and the number of elements that make it up.
To compute the standard deviation an additional value is needed, the
Maksym,
If you can control the Call-IDs that you are receiving, you can prefix
them with something like foo///Call-ID and bar///Call-ID, which are
both treated as having a call id of Call-ID.
Otherwise, you could probably do some hacking to the get_call_id function
so that it will return
December 06). That will give us a bit of time
to stabilize.
Thanks!
Olivier.
On 3/14/07, Charles P Wright [EMAIL PROTECTED] wrote:
Hello all,
I've attached a new patch set with the following patches:
- vgfix.diff
Various valgrind fixes.
- warnings.diff
Allow
Harsimran,
Try something along the lines of the following:
recv request=BYE timeout=timeout ontimeout=1 /
label id=1 /
send
... BYE ...
/send
Charles
[EMAIL PROTECTED] wrote on 03/14/2007 02:09:34 AM:
Hi
How can I design a scenario to have a optional recv before a send
sequence.
What I
Unless the INVITEs have the same Call-ID, they will start a new scenario.
Charles
[EMAIL PROTECTED] wrote on 03/14/2007 04:57:18 PM:
Hello sipp-users,
I need to run sipp with scenario that begins with recv
request=INVITE command
but I do not need to run the scenario from beginning
Anil,
Make sure that your REGISTER and INVITE have the same Call-ID. If you
need to have two different call-id's you can use /// to prefix them with
something. For example: Call-ID: FOO///[EMAIL PROTECTED] and Call-ID:
BAR///[EMAIL PROTECTED] should be treated as the same Call-ID by SIPp.
You can put a NOP before and after the pause with a start_rtd and a rtd,
and then use -trace_rtt.
For example:
nop start_rtd=2 /
pause normal=true mean=100 stdev=10 /
nop rtd=2 /
If you look at the scenario_pid_rtt.csv you can pick out the RTTs with a
Rtd_no of 2.
Charles
[EMAIL PROTECTED]
(2) The update_nb option seems to take great care to avoid calling
getmilliseconds() on every loop. Is there a particular system that
this call is very expensive on?
Olivier,
I've done a quick test on Linux and Windows and found that it is quite
fast (395/390 nanoseconds), certainly
last_recv_hash, then we should increment the
retransmission counter.
Charles
--
Dr. Charles P. Wright
Research Staff Member
Network Server Systems Software
IBM T.J. Watson Research Center
retranshash.diff
Description: Binary data
Paul,
From a quick look at the code, it seems that the peer tag is only picked
up out of responses and not replies, and in your scenario there are no
received replies before you use the peer_tag_param.
You should take a look at this bit of code:
/* It is a response: update peer_tag */
Can someone explain the difference between the main_socket and the
tcp_multiplex socket? It seems that the TCP multiplex socket should only
be used if we are using -t t1 (otherwise each call gets its own socket).
But, why are things sent over the tcp_multiplex socket instead of just the
Peter/Olivier,
I think instead of adding a start_line attribute, that behavior should be
the default (as most people probably expect it to work that way).
Charles
Olivier Jacques [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
02/14/2007 02:15 PM
To
Peter Higginson [EMAIL PROTECTED]
cc
One option that may work for you is to use two separate SIPp instances
(one for registration and another for the UAS behavior).
Charles
[EMAIL PROTECTED] wrote on 02/07/2007 10:41:38 AM:
First I want to thank the maintainers for an incredibly useful tool and
a good job!
Now my question:
.
Can this be done using the same ip:port?
Niklas
On Wed, 2007-02-07 at 10:48 -0500, Charles P Wright wrote:
One option that may work for you is to use two separate SIPp instances
(one for registration and another for the UAS behavior).
Charles
[EMAIL PROTECTED] wrote on 02/07
What release of SIPp are you using? The latest (1.1-rc8) has lots of TCP
fixes, which should hopefully solve your problems.
Charles
[EMAIL PROTECTED] wrote on 01/09/2007 01:26:53 PM:
Hi,
I am using SIPp and SER for some performance study and
get several errors when using TCP protocol.
The problem seems to be this memset line at call.cpp:2196 (or
thereabouts):
memset(my_auth_pass,0,KEYWORD_SIZE);
key = getKeywordParam(src, password=, my_auth_pass);
If you remove that line, the authentication from the command line should
be used.
Charles
[EMAIL PROTECTED]
of changes to TCP handling
in SIPp. Could you explain how SIPp decides if a SIP
message is complete or partial? Since I suspect the
problem I am seeing could be caused by SIP message
compatibility issue between SER and SIPp.
thanks,
Joy
--- Charles P Wright [EMAIL PROTECTED] wrote
Maksym,
The duration is specified in milliseconds, so you should use -d 5000 for a
five second pause.
Charles
[EMAIL PROTECTED] wrote on 12/25/2006 05:13:03 AM:
Hello sipp-users,
I run SIP client with -d command-line option.
sipp ... -d 5 ...
XML scenario contains pause/ string (without
=104305package_id=:
119322release_id=472717
This is the occasion to warmly thank all the contributors that
to this release, with a special thanks to Charles P. Wright
research for a huge set of new features and improvements.
that some fixes and enhancements didn't make their way
especially thinking of FreeBSD
-patches-2006-12-20-ibm1.tar.gz)
Olivier Jacques [EMAIL PROTECTED] wrote on 12/20/2006 10:55:37 AM:
Charles P Wright wrote:
patches to the 2006-12-08 release.
Charles,
I have checked all the changes in.
http://sipp.sourceforge.net/snapshots/sipp.2006-12-20.tar.gz
Thanks a lot, again,
--
Olivier
You can use \x to have a hex-encoded character. Try doing:
INVITE sip:[EMAIL PROTECTED]::214:4fff:fe22:d312] SIP/2.0
This should prevent SIPp from treating the IPV6 address as a keyword.
Charles
[EMAIL PROTECTED] wrote on 12/13/2006 08:08:05 AM:
Hi Friends,
I am trying to use an IPv6
handling of invalid
options. Also use the option table to automatically generate the help
message.
Charles
--
Dr. Charles P. Wright
Software
(See attached file: sipp-patches-2006-12-08-ibm1.tar.gz)
sipp-patches-2006-12-08-ibm1.tar.gz
Description: Binary data
loss.
Charles
--
Dr. Charles P. Wright
Software
(See attached file: retransoption.diff)(See attached file:
countbeforeaction.diff)
retransoption.diff
Description: Binary data
countbeforeaction.diff
Description: Binary data
Hello,
I have attached a few patches with some enhancements and a few bug fixes to
my previous patches. The major enhancement is that TCP reads are no longer
octet-by-octet at the system call level. The bug fixes are for TCP partial
message handling and also pcap playing. I haven't actually
To
Bruno, Guerin (NonHP :
10/20/2006 09:58 AtosOrigin) [EMAIL PROTECTED],
AMCharles P Wright/Watson/[EMAIL
PROTECTED
Hello,
I have a question regarding TCP support under congestion in the latest SIPp
releases. Previously, there was discussion on the mailing list that
pointed out incorrect behavior when the TCP window closed and send returned
EWOULDBLOCK or EAGAIN.
I looked at the send_message function, and
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