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
To
Bruno, Guerin (NonHP :
10/20/2006 09:58 AtosOrigin) [EMAIL PROTECTED],
AMCharles P Wright/Watson/[EMAIL
PROTECTED
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
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
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
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
-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
=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
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
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
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
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
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
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 */
(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
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]
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.
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
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
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
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
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
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?
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
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
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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
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
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
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
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
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
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
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
, 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
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
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
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
=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
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
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
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
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],
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
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
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 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
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 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
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
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
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
. 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
. 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
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
You must use two SIPp instances for this scenario.
Charles
[EMAIL PROTECTED] wrote on 10/03/2007 10:46:40 AM:
can sipp handle the scienario ?
sipp server
| register|(call-id 1)
| |
| 200(OK) |(call-id 1)
| subscribe|(call-id 2)
Your registration message can have a contact address that matches where
you expect the subscribe to go, not necessarily where it originated from.
Charles
Simon Flannery [EMAIL PROTECTED]
10/03/2007 12:09 PM
To
Charles P Wright/Watson/[EMAIL PROTECTED]
cc
yuan [EMAIL PROTECTED], sipp-users
expiry time using Sipp. I get's registered and
remains only for about 30 minutes. Should I have increase the expiry
time at the server or is there any option to increase it in the SIPP
tool.
Best Regards
Karthik.A
-Original Message-
From: Charles P Wright [mailto:[EMAIL PROTECTED
With the simple UAC/UAS scenario I can do 10,000CPS (SIP only, no media)
when using a two processor 2.6Gz dual-core processor AMD Opteron. This is
with two SIPp instances (one acting as a UAC and another acting as a UAS),
which is required because of the single-threaded event driven
If you want to obtain maximum performance you need to increase various
networking buffers using /proc/sys or sysctl.
Charles
[EMAIL PROTECTED] wrote on 10/10/2007 10:43:49 AM:
I have run a peak of somewhere around 5000 signaling only CPS
between servers on a locally switched network.. I
What exactly do you mean by It doesn't do anything?
If you want to debug things, I would remove the -r and -rp options and see
if that does something.
Charles
Bhavin [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
10/10/2007 05:41 PM
To
sipp-users sipp-users@lists.sourceforge.net
cc
Subject
Try find /usr/include -name md5.h and see what comes up.
Charles
[EMAIL PROTECTED] wrote on 10/14/2007 10:49:57 AM:
Hi,
I have changed my Linux distribution to Debian_4.0 and now have a
problem at
compiling SIPp
I have of course installed openssl. But when I compile make ossl I get
One way to help the whole community is to post a sample scenario on the
Wiki:
http://sipp.sourceforge.net/wiki/index.php/Scenarios
Charles
[EMAIL PROTECTED] wrote on 10/19/2007 02:10:10 AM:
Hello Michael,
We, in our testing, have been using PRACK. I assume we have a GOOD
example. Let me
to separate out the refreshing, from the actual call
flow, so that I don't have to worry about refreshing at various points
of my call flows (that can last many many hours).
tia,
rouble
On 10/24/07, Charles P Wright [EMAIL PROTECTED] wrote:
You can do something like:
label id=1 /
send
Michael,
I have never done it, but you do require a few modifications. For the
pause:
pause variable=2 /
You also probably need to multiply $2 by 1000 to convert from seconds to
milliseconds. In your original action add:
multiply assign_to=2 value=1000 / !-- $2 *= 1000 --
One very minor thing
503 Server too busy (containing a
Retry-After-value) and sometimes 503 Service Unavailable (no
Retry-After). Is it possible to differ between these two Responses?
A recv response=503 would catch both of the messages ...
thanks!
BR
Michael
Charles P Wright wrote:
Michael,
I have
Unfortunately, I do not have any examples using 3PCC to coordinate split
flows.
Charles
rouble [EMAIL PROTECTED] wrote on 10/24/2007 05:25:59 PM:
Can you point me to some example scenario files?
tia,
rouble
On 10/24/07, Charles P Wright [EMAIL PROTECTED] wrote:
You'll need to split
Scott,
Thanks for your debugging and analysis, and not just blindly accepting my
explanation. It is easy to read what you think should happen (or intended
to happen) into code that you wrote rather that just accepting.
[EMAIL PROTECTED] wrote on 10/26/2007 11:23:18 AM:
However, for #2 this
There is no support for changing the IP options, but if you can do it
without resorting to raw sockets, you could probably write code to do it
without too much trouble.
Charles
[EMAIL PROTECTED] wrote on 10/27/2007 05:14:45 AM:
Hi
I just want to know that SIPP has any options or
Presently, you can not compare two variables you can only compare a
variable and a value. Also for strings, you need to use the strcmp
primitive instead of the test primitive.
Charles
Sumeet Bhardwaj [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
10/27/2007 07:52 AM
To
optional=true
/recv
recv response=200
optional=false
/recv
/scenario
Please help me out..
Thanks and Regards
SUMEET BHARDWAJ
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-Original Message-
From: Charles P Wright [mailto:[EMAIL PROTECTED]
Sent: Sunday
It is a bug that has been fixed in the latest trunk revision.
Charles
[EMAIL PROTECTED] wrote on 10/29/2007 10:52:25 AM:
Hi to all,
I'm using sipp for testing my own Registrar/Proxy Server. Scenario is
simple SIP call, by Proxy: UA1 calls UA2.
When server proxies INVITE message, recieved
SIPp should answer on port 5060 by default. The other ports like are
used for control and media. To make sure the SIP port is 5060 do sipp -p
5060.
Charles
[EMAIL PROTECTED] wrote on 10/31/2007 08:11:05 PM:
Hi everybody there !!!
I am trying to use SIPp to answer calls with the
You can not compare two variables, and the test operator only works on
doubles.
Charles
[EMAIL PROTECTED] wrote on 10/27/2007 07:38:52 AM:
Hi,
I am using windows SIPp tool to load xml scenario file
But getting some problem with my XML code.
I am trying to compare two string stored into $1
There should be a slash before those paths (unless you are in /
already).
For example:
echo * /dev/udp/localhost/5067
Also, you need to use the correct control port, which by default is ,
but in the trunk versions you can control it with -cp.
If the echo command does not work, you can try
of the world Learn more!
Discover the new Windows Vista Learn more![attachment uasbasico.
xml deleted by Charles P Wright/Watson/IBM] [attachment
netstatoutput.txt deleted by Charles P Wright/Watson/IBM
I do not know how to send DTMF digits with an INFO message, but if it is
just a SIP header or body, then SIPp will support it providing you write a
custom XML script.
If you need to do out-of-band RTP signaling it may be possible as well,
but others will have to speak to that.
Charles
[EMAIL
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Sec Invite Scenario
Content-Length: 0
]]
/send
recv response=200
/recv
/scenario
[attachment jagadesh.munta.vcf deleted by Charles P
Wright/Watson/IBM
Wolfgang,
The combination of 3pcc and executing actions will probably get what you
need, as the exec actions can be any script or program that you write.
Charles
[EMAIL PROTECTED] wrote on 11/08/2007 10:05:10 AM:
Hi Oliver,
I am not quite sure but I think the 3pcc required always sipp
0
200 -- E-RTD1 0 0 0
Regards,
Johnny
[attachment test.csv deleted by Charles P Wright/Watson/IBM]
[attachment uac_g729.xml deleted by Charles P Wright/Watson/IBM
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