SIPp's XML parser is unfortunately not an XML parser, so errors are not
caught very well at all.
The timeout value should be fine. What is the precise error message you
see (please copy and paste). Any extra spaces or things would mess up
this value.
If you want help getting the scenario
The -m should count both successful and failed calls.
Charles
paroxyzm [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
05/14/2008 07:47 PM
To
cc
sipp-users@lists.sourceforge.net
Subject
[Sipp-users] Number of calls question
Hi!!
I have a question:
Is there a way to limit number of
, ontimeout
attribute)
What is going wrong, a reference to such a label does not exist?
Thank You, Wolfgang
Deutsche Telekom AG
Zentrum Technik Einführung
Wolfgang Kanngießer
Winterfeldtstraße 21, D-10781 Berlin
[attachment reg.xml deleted by Charles P Wright/Watson/IBM
If you can generate the call-IDs yourself, then you can do something of
the format: prefix///call-id and prefix1///call-id and SIPp will treat
them as the same scenario. Otherwise, you would need to extend SIPp.
Charles
Philip Campion [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
the
cumulative limit, I mean that each one of the instances reach 14,000
open udp sockets and then the uac instance was thrown with the same
error...
Thanks for the reply.
Igor.
-Original Message-
From: Charles P Wright [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 22, 2008 23:21
I am not sure that this is the case, but I suspect that you are running
into OS limits. You may want to try multiple SIPp instances each bound to
different IP addresses. Not sure if that will resolve your issue or not
though.
Charles
[EMAIL PROTECTED] wrote on 04/22/2008 09:13:13 AM:
Hi,
1. The Call-ID must match.
2. There is no reason it shouldn't as long as the call-id matches, and
then the CSeq matches.
Charles
[EMAIL PROTECTED] wrote on 04/14/2008 09:51:41 PM:
I am doing some testing which requires to run sipp in two mode, one with
INVITE and another with INFO. First,
You should look at the regexp action.
Charles
Niven N [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
04/08/2008 10:56 AM
To
sipp-users@lists.sourceforge.net
cc
Subject
[Sipp-users] Checking on SDP information - Help needed
Hello ,
I am try to check if an end to end call was
Bharath,
The situtation you describe seems more like something you would want to do
with a user-based benchmark. That is you have N open calls at any one
point in time to reflect a given population of users.
Aside from that, it seems there is probably a pause missing between the
reinvite and
taking over the
control flow by doing advance_state=false, but I would prefer to fix the
matching for your scenario.
Charles
Scott Oaks [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
04/03/2008 12:54 PM
Please respond to
[EMAIL PROTECTED]
To
Charles P Wright/Watson/[EMAIL PROTECTED]
cc
sipp
My best guess is that if you look at the default messages in the source,
that ACK will match up relatively nicely to one of them in there rather
than it coming from some sort of corruption.
I would also be interested if you can provide any more information about
regressions you've seen.
Anuj,
I think that a 3.1 release would be a very good idea. I have started
documenting the changes. I am have just committed the changes for
387-398. In the coming days I will try to work my way through to the
most recent revision.
Charles
Srivastava, Anuj Kumar [EMAIL PROTECTED] wrote on
regards,
/Christer
-Original Message-
From: Christer Skeppstedt
Sent: Tue 2008-03-18 14:16
To: Charles P Wright
Cc: sipp-users@lists.sourceforge.net;
[EMAIL PROTECTED]
Subject: RE: [Sipp-users] Segmentation fault when 2 video codecs
Hi,
I'm a bit new to this but I can't find
Run make clean ; make ossl and try again.
Charles
[EMAIL PROTECTED] wrote on 03/25/2008 09:42:32 PM:
Hi Simon,
Thanks for your response.
I modified the spelling and still getting error unsupported keyword
authentication in xml scenario file.
I tried to generate the registration for valid
Michael,
You may be able to do something by creating 3 instances of SIPp and third
party call control, in which the first instance sends to the session
director, and the second and third send to the other IPs.
Charles
[EMAIL PROTECTED] wrote on 03/19/2008 09:26:23 AM:
All,
I have a
|
|
uas
For that I want to add Route: header to the invite.
The problem is with the uac. It sends the invite with IP address of the
UAS.
On Tue, Mar 18, 2008 at 2:48 PM, Charles P Wright [EMAIL PROTECTED]
wrote:
SIPp won't use any routing headers unless you
Orante,
You need to modify the built in XML. You can get a template to modify by
running:
sipp -sd uac new-scenario.xml
Try searching on google for sipp register xml.
Charles
Orante Tucceri [EMAIL PROTECTED] wrote on 03/17/2008 07:48:14 AM:
Hi,
i would execute test about Asterisk
Neil,
I haven't ever used pcap, so I have no experience with your issue. The
Aborted looks like SIPp might be getting a signal that is killing it. Try
running it through gdb to see what is causing the failure.
Charles
[EMAIL PROTECTED] wrote on 03/17/2008 04:18:29 AM:
Take 2 J
Hi I have
Bharath,
With the newest trunks you should be able to do this without any options.
If you can't, then the -mp option is the only way to do it.
Charles
[EMAIL PROTECTED] wrote on 03/12/2008 07:10:30 PM:
Hi,
I am trying to run 4 UAC sipp processes on same machine with
different
Please use [EMAIL PROTECTED] for this type of question.
I don't understand the question, but if you do -s 1234 then SIPp will
address the call to [EMAIL PROTECTED] host].
Charles
orante tucceri [EMAIL PROTECTED] wrote on 03/11/2008 07:50:03 AM:
Hi,
from reading the log's file only invite
to resolve?
Best regards
[EMAIL PROTECTED]
Inviato da Yahoo! Mail.
La web mail più usata al mondo. http://it.docs.yahoo.
com/mail/overview/index.html
[attachment test.doc deleted by Charles P Wright/Watson/IBM
Davide,
I don't know why you are getting the unexpected messages, that is likely
based on your OpenIMS configuration.
I have seen incorrect checksums when tracing SIP traffic with Wireshark;
but they have never caused me problems. SIPp isn't involved in the UDP
checksumming, and my guess is
I'm not sure I like just ignoring this type of error. Maybe you can make
it a command line option, or at the very least fail the call?
Charles
[EMAIL PROTECTED] wrote on 03/05/2008 05:13:55 AM:
Hello all,
I'm sending this mail to the sipp user list as apparently there is
not much life on
What version are you using?
Charles
[EMAIL PROTECTED] wrote on 03/05/2008 08:29:13 AM:
Hi,
I am the starter to use the Sipp. Now I have a problem.
When I finished installing the Sipp and ran the command ./sipp -sn
uas, it is ok.
But in the same host , when I ran the command ./sipp -sn
This is a bug, but I don't know anyone else who is came across it. If you
run ./sipp through gdb can you produce a backtrace?
Charles
Xiaotian Shen [EMAIL PROTECTED] wrote on 03/05/2008 09:05:57 AM:
3.0
To: [EMAIL PROTECTED]
CC: sipp-users@lists.sourceforge.net; [EMAIL PROTECTED]
I was pretty sure this was fixed. Can you send a simple set of XML
scenarios (UAC and UAS) that replicate this bug?
Any bug of this nature is most likely in get_last_header, but
CreateSendingMessage could be at fault as well.
Charles
[EMAIL PROTECTED] wrote on 03/05/2008 09:10:50 AM:
Hey
The easiest way to get this done would be to use the -users option. This
won't let you use the '+' and '-' keys right away, but it would get you
one step closer (and you can probably modify the meaning of '+' and '-'
when using the -users mode).
Charles
[EMAIL PROTECTED] wrote on 02/28/2008
You should have separate ports for 3pcc and SIP traffic. Also, I think
you should have the same address in both 3pcc arguments; it should be the
address of the instance that has recvCmd first.
Charles
[EMAIL PROTECTED] wrote on 02/28/2008 09:31:37 AM:
Hi all,
I'm having a problem when
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Charles P
Wright
Sent: Tuesday, February 26, 2008 9:37 PM
To: Niven N
Cc: sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] Conditional Brancing using test
Niven,
I am glad that this works for you. I am
The last_cseq_number is taken from your last response not the messages in
the scenario.
Charles
Fabio Margarido [EMAIL PROTECTED] wrote on 02/26/2008 09:42:07
AM:
On Fri, Feb 22, 2008 at 3:41 PM, Charles P Wright [EMAIL PROTECTED]
wrote:
In the trunk revision you can change the built
The [cseq] keyword might work for you (if you use it in all of your
messages). Alternatively, you can introduce an integer variable and
increment it yourself for the CSeq.
Charles
Fabio Margarido [EMAIL PROTECTED] wrote on 02/26/2008 11:40:51
AM:
On Tue, Feb 26, 2008 at 12:14 PM, Charles P
I have tested all the possible cases and it works fine.
Thanks again !
Niven.
On Tue, Feb 26, 2008 at 10:16 AM, Charles P Wright [EMAIL PROTECTED]
wrote:
Niven,
Try inserting a todouble after the assignstr and see if that makes a
difference. You won't be able to properly compare
Evgeny,
I quickly checked the source and it seems regular expression matches only
work for requests. If no one has any other ideas, my suggestion is that
you may be able to modify the call:matches_scenario function and scenario
parsing in scenario.cpp to add support for responses being
You want something like:
recv response=183 optional=true next=5
/recv
recv response=180
/recv
[EMAIL PROTECTED] wrote on 02/22/2008 07:31:19 AM:
Hi all,
I want SIPp to jump to the label when it is expecting 180 response
but it gets 183 response from UAS
I have tried many combination to
In the trunk revision you can change the built-in messages with the
DefaultMessage element:
DefaultMessage id=bye
![CDATA[
BYE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
X-Extra: CPW
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
I'm not 100% sure, but my best guess is that SIPp may not be handling the
CSeq with a space correctly, but I don't know for sure.
Charles
Chris Cunningham [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
02/20/2008 05:05 PM
To
Sipp-users@lists.sourceforge.net
cc
Subject
[Sipp-users] Aborting
You should use assignstr instead of assign.
Charles
[EMAIL PROTECTED] wrote on 02/19/2008 09:19:18 AM:
Hi all,
I m using regular expression to verify whether particular number is
present in ?To? header field or not.
The number I m providing from CSV file to the regular expression.
But
The peer_tag_param should pull the To: tag from the 200. You can probably
get the From tag using a regular expression.
Charles
[EMAIL PROTECTED] wrote on 02/19/2008 08:48:50 AM:
Hi All
I'm running SIPp in server mode, with a scenario as follows:
-- SUBSCRIBE
-- 200 OK
-- NOTIFY
-- 200
There is no pre-defined limit, but you can run out of memory if the file
is too large.
printf files are designed to provide larger files by letting you create
virtual lines by substituting the line number into the text. For example:
RANDOM,PRINTF=100
user%04d
Will create a file that virtually
Changing the code to support it should be fairly straight forward. Just
search for NOTIFY in the code.
Charles
[EMAIL PROTECTED] wrote on 02/06/2008 10:13:24 AM:
Hi,
I am try to enable on SIPP an automatic 200 OK answer for OPTIONS
message, but I checked the -aa just enable auto aswer
I don't know what you mean by real-time media, but as far as I know SIPp
only supports sending a pcap file or echoing back RTP data that was
received.
Charles
Nobody [EMAIL PROTECTED] wrote on 02/01/2008 06:41:54
AM:
Message body follows:
hi charles,
does sipp support sending real
I checked the 3_0 branch and it looks like it should be there.
Charles
[EMAIL PROTECTED] wrote on 01/30/2008 03:28:19 AM:
Thanks...
But is '-no_rate_quit' option available in sipp 3.0? I can't find in
docs or in source? sipp complains saying Invalid argument:
'-no_rate_quit'
-Bharath
Bharath,
Yes you can do this with the fllowing options:
./sipp -r n -rate_increase x -fd p -rate_max N -no_rate_quit
The value of p will control both the statistics interval and also the
interval between increases of x calls per second.
Charles
[EMAIL PROTECTED] wrote on 01/29/2008 09:14:43
,
Frédéric-Philippe Metz
On Fr, 2008-01-25 at 09:30 -0500, Charles P Wright wrote:
Frederic-Phillippe,
The 3.0 code has much improved message parsing (it catches many errors
that were not otherwise caught and greatly improves performance);
unfortunately it does not work
at 10:29 -0500, Charles P Wright wrote:
The -timeout option will simply cause SIPp to exit after a certain
amount
of time. The -recv_timeout option is applied only to messages that do
not
have an explicit timeout specified in the XML.
The first timeout is taken for a sequence
The -timeout option will simply cause SIPp to exit after a certain amount
of time. The -recv_timeout option is applied only to messages that do not
have an explicit timeout specified in the XML.
The first timeout is taken for a sequence of messages. As SIPp doesn't
have a notion of
Frederic,
The timeout XML attribute is for each particular message, as calls are not
created until the first message is received, there is never an opportunity
for the timeout to be applied to the first message (as by definition a
message would have been received for the call to exist). It
You have a CSeq of 1 INVITE, when it should be 1 SUBSCRIBE. This might be
confusing the transaction matching algorithm.
Charles
[EMAIL PROTECTED] wrote on 01/25/2008 08:06:39 AM:
Hi
I'm using SIPp to test a program based on the eXosip2 SIP stack, which
automatically sends a 101 Dialog
SIPp is very configurable and can therefore support any SIP method.
Charles
[EMAIL PROTECTED] wrote on 01/17/2008 04:59:12 AM:
Hi, all
I am a new user of sipp. Now I want to use this tool to test SIP
UPDATE method,but I am not sure if this tool support update method
or not.
Can you give
Ding,
I do not know about Windows (from what I've heard it is much less), but on
Linux the answer is 1000s of CPS depending on your exact scenario.
Charles
Ding, Peng [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
01/16/2008 05:01 AM
To
sipp-users@lists.sourceforge.net
cc
Subject
You can't use expansions inside of your regular expression. The best bet
would be for you to checkout the trunk branch, create a regex to extract
the number, use the assignstr action to store the key (which shoudl be
[key] not [$key]), and then modify the strcmp action code to accept two
If you have a start_rtd XML attribute you must have a corresponding RTD
attribute.
Charles
Maria Rosa Vieira Alvarez [EMAIL PROTECTED]
01/09/2008 10:25 AM
To
Boulkroune, Olivier (Non-HP:Atos Origin) [EMAIL PROTECTED],
Charles P Wright/Watson/[EMAIL PROTECTED]
cc
sipp-users
Varun,
When HP implemented SIGCOMP support they did it through a plugin
architecture. Unfortunately, not compression plugins are released.
Compression is only supported for UDP.
There is a pretty simple interface that would need to be implemented
consisting of three functions:
int
user has an IP
address) to SIP proxy?
2. If SIPp 3.0 can support TCP with multiple socket for both client and
server, please show me the way to run it.
I appreciate your help.
Best regards,
Vinh
-Original Message-
From: Charles P Wright [mailto:[EMAIL PROTECTED]
Sent
Without the exact error message you are getting, it is almost impossible
to teel what is wrong.
Charles
[EMAIL PROTECTED] wrote on 12/28/2007 04:41:30 AM:
Hi
I need help regarding to call transfer scenario (A calls B, and B
user transfers call to C user). I?ve understood is needed to
Roman,
This is not possible with a single SIPp instance, but can be accomplished
using third party call control. Essentially, the call flow would have to
be broken up into the following:
UAC1 - Sends INVITE, recieves 180 and 200, sends a command message to
UAC2, instructing it to send an ACK
in the error log file after this one?
Charles
K L [EMAIL PROTECTED] wrote on 01/02/2008 11:22:46 AM:
On 1/2/08, Charles P Wright [EMAIL PROTECTED] wrote:
K L,
If you jump to the end of the call SIPp marks it as failed
automatically
(which I think is probably not the best possible behavior
Do you have any actions, if not I am not sure why the call would be marked
as rejected without any other messages in the logs. Can you post your
scenario XML?
Charles
K L [EMAIL PROTECTED] wrote on 01/02/2008 01:32:23 PM:
On 1/2/08, Charles P Wright [EMAIL PROTECTED] wrote:
I don't think
Stefan,
I fixed this in the trunk after you reported it. I have backported the
fix to the 3.0 branch as well so that when there is a future 3.0 release
the fix will be included.
Charles
[EMAIL PROTECTED] wrote on 11/28/2007 06:41:51 AM:
Hi,
during tests to understand the features of a
Eliot,
I can't think of any reason that SIPp would read a differnet packet off
the network than what it recieves. Maybe there is some confusion with the
scenario or something.
Can you include the XML and a -trace_msg log with a single single call
(the -m 1 flag will limit SIPp to generating
Average response times are actually measured for successful transactions
not calls (e.g., if your BYE fails, the INVITE response time is still
recorded).
A call can fail for more than one reason, but the most common two are (1)
there was a retransmission timeout, (2) there was an unexpected
Mikkel,
SIPp does support retransmissions, but you need to make sure that your
send line has the retrans=500 parameter (as in the default UAC
scenario). If that still doesn't work, you should post your XML
scenarios.
Charles
mikkel [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
12/06/2007
PROTECTED]
12/06/2007 02:22 PM
Please respond to
[EMAIL PROTECTED]
To
Charles P Wright/Watson/[EMAIL PROTECTED]
cc
sipp-users@lists.sourceforge.net
Subject
Re: [Sipp-users] Refresh Requests using SIPp
Comments inline...
1. Does SIPp ( 2.1 or 3.0 version) support sending refresh requests
Dan,
If you use the latest SIPp (3.0), the branch should already include the
PID:
dest += snprintf(dest, left, z9hG4bK-%u-%u-%d, pid, number,
P_index + comp-offset
Also, you can use [branch-N] (where N is the offset between the two
messages), if you want to make sure that an ACK has
I just want to make people aware of some changes that have been committed
to the trunk since the 3.0 release (r371):
- You can use names for variables and labels.
- You can use keywords for authentication parameters.
- TCP reconnection behavior is more fine grained, so that not all
connections
Stefan,
I am not aware of this issue, but I guess it is not all that surprising if
somehow putting the getch function is returning some sort of error, which
SIPp does not handle.
I was not able to reproduce this (when I tried running SIPp in the
background it would get stopped), however you
--- Charles P Wright [EMAIL PROTECTED] wrote:
If you modify the SIPp source code to use a
SendingMessage structure
instead of a string for the play_pcap_audo element
of the scenario you can
do this. The changes would need to be made in
scenario.cpp, scenario.hpp
If you modify the SIPp source code to use a SendingMessage structure
instead of a string for the play_pcap_audo element of the scenario you can
do this. The changes would need to be made in scenario.cpp, scenario.hpp,
and call.cpp.
Charles
[EMAIL PROTECTED] wrote on 11/26/2007 07:12:19 AM:
No, unfortunately SIPp does not actually understand the SIP headers that
control transport like Via, Contact, and Record-Route, but rather simply
replies to the address where the message came from.
Charles
[EMAIL PROTECTED] wrote on 11/26/2007 08:05:49 AM:
Hi!
I found out that sipp in a
SIPp does support multiple sockets per client (each with a different
port), and even multiple IP addresses, though 1,000 virtual IP interfaces
on a single machine may be stretching its capabilities. I would think
that you may at least need multiple SIPp instances (likely due to file
If you need a thread per call, it is likely that there is an OS limit to
the number of threads that you can use. I suggest trying to run multiple
SIPp instances, and if that doesn't work you may possibly need to run SIPp
on multiple machines. (Unless of course anyone else has a better
Scott,
Change your label to label id=5 /. The situation you are seeing is
that the label is not getting found, so the call ends and a new one
replaces it.
Charles
[EMAIL PROTECTED] wrote on 11/20/2007 01:39:14 PM:
I need to construct a scenario where a sipp uac sends a SUBSCRIBE
message to
wouldn't be so confused
about what's going on.
If there's no reason not to do that, I can generate a patch.
-Scott
On Tue, 2007-11-20 at 13:44, Charles P Wright wrote:
Scott,
Change your label to label id=5 /. The situation you are seeing is
that the label is not getting found, so
?.
-Original Message-
From: Charles P Wright [mailto:[EMAIL PROTECTED]
Sent: Friday, November 16, 2007 7:08 PM
To: Thekkedath, Sooraj (Sooraj)
Cc: sipp-users@lists.sourceforge.net;
[EMAIL PROTECTED]; Thekkedath, Sooraj (Sooraj)
Subject: Re: [Sipp-users] Regarding SIPP performance
Are you using
Are you using RTP? If not, SIPp should have no problem at all handling
100 CPS on anything reasonable (even an old Pentium). I don't know what
the performance is with RTP.
Charles
[EMAIL PROTECTED] wrote on 11/16/2007 07:54:12 AM:
Hi all ,
I was trying to do a performance testing with
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
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
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
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
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
optional=true
/recv
recv response=200
optional=false
/recv
/scenario
Please help me out..
Thanks and Regards
SUMEET BHARDWAJ
| Mob: +919970159464 |
-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
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
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
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
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
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
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
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
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
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
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
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