You need 'auth="true"' on the 407 (i.e. ) so it saves off the challenge data.
On 31/08/16 23:30, Ned Ludd wrote:
>
> Can you explain this error?
>
> Attached is my scenario file.
>
> Thanks.
>
> -
> Ned Ludd
>
>
> --
The way to do this is to customise the XML scenario file -- the default
one uses sip:[service]@[remote_ip]:[remote_port] as the Req-URI.
To use a custom one, just run sipp -sd uac > myuac.xml to dump it to a
file, edit myuac.xml to set the Req-URI you want, then run sipp -sf
myuac.xml [other args]
'-r 15 -rp 25s' means "15 calls every 25 seconds" - pretty close to
one call every 2 seconds. Just '-r 15' should generate a steady 15
calls/second, if that's what you need.
Hope that helps,
Rob
On 12 May 2015 at 16:05, Email Lists wrote:
> Hello,
>
> I'm wondering if anyone has run into this is
estructor, or type
> conversion before ‘(’ token
> src/actions.cpp:775: error: expected constructor, destructor, or type
> conversion before ‘(’ token
> make: *** [src/sipp_unittest-actions.o] Error 1
>
> It seem's gtest/gtest.h is not found...
> If I take a look on this directo
Hi all,
I've just tagged the SIPp 3.5.0-rc1 release candidate
(https://github.com/SIPp/sipp/releases/tag/v3.5.0-rc1).
I'll test it and build it on Fedora Linux, Cygwin and Mac OS X before
declaring 3.5.0, but if you want to build it, ensure it works with your
test scripts, etc., please let me kno
should be:
i.e. the closing tags are wrong.
On 26 February 2015 at 06:43, vinod kunchanur wrote:
> Hi All,
> I am creating a scenario where SIPP will call to an extension and that
> extension will get answered manually and trying to play a media file which
> is in .wav format
>
> Bu
Can you send the full message log?
On 4 February 2015 at 07:22, Paul Miller wrote:
> Hi,
>
> I am using SIPp v 3.4.1, and have the snippit below in a UAC call
> originating script.
>
> =
>
>
>
>
>
>
>
>
>
>
>
> =
>
> As you
trace which shows
> the spamming in the flow diagram. I also have reconnected the phone
> and tried calling the same number to see if it was a network
> problem but that seemed to work fine,
>
> Kind regards,
>
> Alex
>
> On Thu, Dec 25, 2014 at 9:47 AM, Rob Day wro
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
If you have Wireshark traces (or the -trace_msg output file), that
would be useful. From a quick skim of the XML, it looks like it should
work (ACK to the 401 in the INVITE transaction, ACK to the 200 OK in a
new transaction), but maybe something unexp
nction it
> appears in.) ./pcap-dbus.c: In function ?€˜dbus_activate?€™:
> ./pcap-dbus.c:165: error: ?€˜DBUS_ERROR_INIT?€™ undeclared (first
> use in this function) make: *** [pcap-dbus.o] Error 1
>
> Do you know how to solve this issue?
>
> Thanks, Michael
>
>
>
SIPp uses extended POSIX regular expressions, which are pretty common.
http://regex.larsolavtorvik.com/ lets you test them.
Modifying the XML you posted the other day slightly, should do what you want, where the key
thing I'm doing is throwing away the main match (that's the _ in
assign_to, and t
ll actions in a block execute.
Best,
Rob
On 4 October 2014 00:28, Paul Miller wrote:
> Hi,
> I am using the microsip client to originate the call and do not know how (if
> it's possible) to stop the REGISTER attempts so I have been ignoring them.
> I have attached two logs files
Wireshark trace or SIPp's -trace_msg log would be useful, I think, in
order to see the PRACK that's being matched.
On 3 October 2014 23:28, Paul Miller wrote:
> Hi Volkan,
>
> It caused a mall formed packet alert in wireshark and shows like this:
>
> Status-Line: SIP/2.0 200 OK
> Via: SIP/2.0/UDP
Hi Paul,
This email and both your previous ones reached me (and are visible at
http://sourceforge.net/p/sipp/mailman/sipp-users/). I'll probably take
a look over the weekend.
Best,
Rob
On 3 October 2014 20:39, Paul Miller wrote:
> I have been trying for almost two days to send a new message to
Hi Paul,
A timestamp in seconds for theTimestamp header can be created by using
the gettimeofday action -
http://sipp.sourceforge.net/doc/reference.html#gettimeofday.
SIPp 3.4.1 had no support for the date, but it looks like a useful
feature, so I've just added it to the codebase at
https://githu
Looks like the valid form is just "From:
;tag=fmTag-1234567890-0". From the
part you quote, user-unreserved is a valid component of userinfo, and
',' is a character in the user-unreserved set, so it can be used
directly without escaping.
For reference, escaped characters are not done with backslas
On 21/07/14 23:43, Praveen Bandari wrote:
> Hi All,
>
> Kamailio developers in my organization have implemented a Load
> Balance using DNS with two Kamailio servers behind. This setup is
> like as below
>
> -> When the client (mobile) sends a request to server.com -> The
> DNS will either resolve
On 21/07/14 23:56, Praveen Bandari wrote:
> Thanks for the reply. Actually the problem is with the SIPp, but I
> might be wrong here. Could you validate this.
>
> -> When I register the user1 on server with -t tn and without
> providing the port, the location table shows clientIP:45128 (UAC)
> ->
On 19/07/14 02:40, Paul Miller wrote:
> I was thinking I could something like this:
>
> ==
>
>assign_to="authvalid" username="1000" password="1000" /> message=" authvalid=[$authvalid]"/>
>
> assign_to="auth" variable="authvalid" />
>
>
>
> but I get this error
Hi Paul,
Using your files, I'm able to reproduce the issue exactly on v3.4.0,
but not on v3.4.1 - which is as I'd expect, since I fixed a bug here
in v3.4.1. You previously said you were seeing this on v3.4.1, but I
looked back through the email trail, and this is the version string
you quoted:
"
Hi Praveen,
On 15/07/14 09:32, Praveen Bandari wrote:
> I am receiving a 477 message from SIP server while sending a
> message to a registered users using SIPp. Did not see the message
> when I send the message to a registered user from iPhone client.
>
> Could you please let me know if you have
On 15/07/14 16:43, Norman Wagner wrote:
> After a while (1h to 10h) SIPp crashes with following error
> message:
>
> sipp: src/task.cpp:172: task_list* timewheel::task2list(task*):
> Assertion `wheel_base <= clock_tick' failed. ./inv_with_auth.sh:
> line 14: 30059 Aborted (core du
On 09/07/14 13:19, Mahudeswaran A wrote:
> Hello,
>
> once the max retrans count is reached what will be the SIPp
> behaviour?
>
> [Ans] after the max retrans count is reached SIPp will stop sending
> further INVITEs and that call is maintained in “Current Call”
> count;
>
> --Is my understandin
On 13/07/14 04:44, Praveen Bandari wrote:
> Hi All,
>
> I have requirement to perform REGISTER method for every 5 minutes.
> Right now, I am doing this using a loop created with the use of
> 'label', but this is causing too many requests being sent to
> server.
This is the way I would do it. I'm
On 11/07/14 05:49, Paul Miller wrote:
> is there a way to log what response that verify auth was expecting
> so I can troubleshoot further? Or any further troubleshooting
> pointers would be awesome.
>
> Thanks Paul
>
If you use the -trace_calldebug argument, you'll get this log:
https://github.
On 11/07/14 19:03, Joe Schmid wrote:
> Hello,
>
> I'm trying to have rtp_stream send a different audio file for each
> call using SIPp v3.4-beta1 (aka v3.3.990)-SCTP-PCAP-RTPSTREAM. I
> see that my variable is getting the expected filename string, but
> rtp_stream doesn't appear to be processing
What version of SIPp are you using? I've made some fixes to verifyauth
in 3.4.1 (which may have fixed your bug, or which may have caused a
regression).
Best,
Rob
On 9 July 2014 10:22, Paul Miller wrote:
> Hi all,
> To try and find a successful baseline that shows me that verifyauth is
> working
On 30/06/14 05:27, Patgar, Geeta Ganapayya (STSD) wrote:
> Hi,
>
> We are trying to build sipp-3.3.990 on hpux 11i v3 IA64 system, but
> getting below errors, any idea what might have gone wrong?
>
>
> /usr/local/lib/gcc/ia64-hp-hpux11.31/4.2.3/../../../../include/c++/4.2.3/fstream:
> In member
On 01/07/14 14:40, Tomasz Karbowski wrote:
> Hello,
>
> I tried to use chance in my UAS scenario for getting 50% calls
> connected and 50% calls rejected by 486 Busy here.
>
> I am getting all time 486 busy here instead of 50% busy here and
> 50% 200OK to connect the call.
>
>
> Part of XML UAS
On 18/06/14 06:35, Sambhaji Gayake wrote:
> As you can see, it has Accept-Contact header before Contact header.
> Now if I use below expression to extract Contact header, it always
> gives me Accept-Contact value.
>
> assign_to="tempFarContact"/>
>
>
> Is there a way to make sure we get Contact
On 04/07/14 23:44, Paul Miller wrote:
> Also I am using check_it=true on
> my regex lines but even if they fail to match the test does not exit
I think that might be due to https://github.com/SIPp/sipp/issues/53,
where re-using a match variable causes incorrect successes - as a
workaround, can you
On 26/06/14 15:47, Baert Valentin wrote:
> Hello
>
> Does someone has managed to compile SIP 3.4 under Cygwin and if yes
> can you tell me how you did it please, because the SIPp
> documentation is not helping very much about that. I've visited
> many website but there is no solution working for m
On 02/07/14 19:31, Filip Planinsky wrote:
> Hi,
>
>
> In order to fix the problem with the Terminal on MAC OS X, you
> should add setlinebuf(stdout);
>
> At the beginning of print_statistics() in logger.cpp
Thanks! I'll get this change made and the usleep removed. (I suspect
the usleep works be
On 04/07/14 06:05, Paul Miller wrote:
> Hi all, I am sure I am doing something stupid but I cannot seem to
> get the Log Message Working, the Warning Message command is working
> great both on the ncurses UI and in the error log.
Hi Paul,
There's a -trace_logs option which is needed to print the
gt;
>
> Thanks & Regards
> Mahu
>
> -Original Message-
> From: Volkan KUMBASAR [mailto:kumba...@netas.com.tr]
> Sent: 11 June 2014 15:23
> To: Mahudeswaran A; Rob Day
> Cc: sipp-users@lists.sourceforge.net
> Subject: RE: [Sipp-users] sipp not gener
On 05/06/14 21:35, Mahudeswaran A wrote:
> Hi Rob,
>
> We just restarted SIPp with the attached scenario file with retrains
> attribute; and snap shot taken so far;
>
> We have monitor for some more time…
>
> Any comments…
>
> [cid:image001.png@01CF812A.7B1FE0D0]
>
>
>
> Thanks & Regards
>
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 05/06/14 21:13, Mahudeswaran A wrote:
> The response time repartition table was also not present. shall we
> add them too.. value="10, 50, 100, 500, 1000, 5000, 1"/>
>
> can you please explain where & when these repartition values will
> be
he ‘Retrans’ keeps
> increasing; how is ‘Retrans’ plays a role in receive end!!
>
> Thanks & Regards
> Mahu
> From: Mahudeswaran A
> Sent: 05 June 2014 12:21
> To: 'Rob Day'
> Cc: sipp-users@lists.sourceforge.net
> Subject: RE: [Sipp-users] sipp not gener
kday/jsipp/wiki/ZeroMQ#rtcp) - but this is not at
all mature yet, and I haven't had time to update it for a few weeks
now.
On 4 June 2014 19:50, Rob Day wrote:
> Can you clarify what you mean by "moss" and "r factor"? I've not come
> across those t
Can you clarify what you mean by "moss" and "r factor"? I've not come
across those terms before, and a quick Google search was not very
enlightening.
SIPp doesn't currently calculate jitter stats, and there's no baked-in
Matlab support - although it makes various statistics available in CSV
format
ut if you look at uac scenario file sipp has to send BYE after 1
> minute...but BYE was not sent...
>
> Regards
> Mahu
>
> Rob Day wrote:
> Your screenshot shows that 100 calls are currently in progress, and the
> "-l 100" command-line argument means that there can&#
Your screenshot shows that 100 calls are currently in progress, and the
"-l 100" command-line argument means that there can't be any more than 100
calls open at a time, so no new calls will be opened until some existing
calls end.
Rob
On 29 May 2014 20:34, Mahudeswaran A wrote:
> Hello,
>
>
On 28/05/14 19:04, Alejandro Galasso wrote:
> Hi All
>
> This is the first time I am using SIPP (version 3.4.1) and I like to
> make an stress test to a lab equipment. I have a problem that is as
> follows:
>
> 1. Client send first INVITE and get 407.
> 2. After that, client send second INVITE(cs
On 25/05/14 08:15, Paul Miller wrote:
Using the following regex in the website I get my desired outcome:
.*:([0-9]{4,5})[:;]
I am sure that this will not be the best way but certainly it works in
the website just fine.
however when I use it an SIPp xml file I never get the same match as
the
Is the problem that you specify "-m 1", causing SIPp to only make a
single call and then exit? Does it work if you use "-m 100"?
Best,
Rob
On 24 April 2014 18:18, .. wrote:
> Hi
>
> I am a new user to sipp and have a problem with the SEQUENTIAL command being
> ignored in a CSV fie.
>
> ../sipp -
Hi Mahu,
SIPp won't compile in Visual Studio as it uses features not supported in
pure Windows. You'll need to use Cygwin.
A while ago, Polycom contributed a variant of SIPp which does compile on
Windows (https://github.com/SIPp/polycom-sipped), but I haven't used this
much, or made much progress
I ran xmllint over it and compared it to the results of 'sipp -sd
uac'. It looks like the of the 100 Trying is copied twice, and
you're missing the "" at the start.
Rob
On 11 April 2014 18:28, Munzer,David J wrote:
> Dear Rob,
>
> I made your recommended changes and checked for any missing
> r
You may need to install 'autoconf' or 'autotools' as well.
Best,
Rob
On 11 April 2014 08:40, Chwolka, Peter (Peter)
wrote:
> Thanks Johan.
>
>
>
> I reinstall automake package 1.13.4-1
>
>
>
> No change of Behaviour.
>
>
>
> Any new idea?
>
>
>
> Thanks
>
> Peter
>
>
>
>
>
> From: Johan Vikman [
Forgot to cc the list.
-- Forwarded message --
From: Rob Day
Date: 9 April 2014 23:24
Subject: Re: [Sipp-users] Connecting phone system to SIPp
To: "Munzer,David J"
David,
It looks like your scenario isn't valid XML - the '
' section appears twic
g.
>
>
> Thankfully,
> David
>
>
> -Original Message-
> From: Rob Day
> To: Munzer,David J
> Cc: sipp-users
> Sent: Tue, Apr 8, 2014 2:11 pm
> Subject: Re: [Sipp-users] Connecting phone system to SIPp
>
> Hi David,
>
> Do you know the SIP URIs you
one of the actual registered users? The
> authorization page did not clarify that well.
>
> Thankfully,
> David
>
>
> -----Original Message-
> From: Rob Day
> To: Munzer,David J
> Cc: sipp-users
> Sent: Tue, Apr 8, 2014 3:04 am
> Subject: Re: [Sipp-use
essage for Call-Id
> '106-3288@10.0.0.210': while expecting '100' (index 1), recei
>
>ved 'SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 10.0.0.210:5060;branch=z9hG4bK-3288-106-0;rport=5060
> From:
Hi all,
I've just released jSIPp 0.0.5. This release adds the "-rtp_sink"
command-line option. When this option is set, each call opens an
associated UDP socket, processes RTP packets coming in on that port, and
makes statistics on jitter, packet loss and out-of-sequence packets
available over the
Yes, that should be enough - SIPp handles SIP Digest authentication by
itself without requiring regexps.It might be worth logging in with an
ordinary softphone (Jitsi, X-Lite, Blink) just to make sure that your
username and password are correct, before you start trying to log in
with SIPp.
Best,
R
Hi Milos,
http://sipp.sourceforge.net/doc/reference.html#SIP+authentication has
the documentation on this feature, and an example. Does that give you
what you need?
Best,
Rob
On 3 April 2014 16:12, Crnjanski Milos wrote:
> I have some problems with registration...
>
> First I need to understand
Hi Milos,
If you use the -trace_err argument to SIPp, you'll get a log file
containing the exact unexpected message it saw. It may be that your
authentication is wrong and so you're getting a 403 Forbidden instead
of the 200 OK.
Best,
Rob
On 3 April 2014 15:09, Crnjanski Milos wrote:
> Dear All
in
> socket, errno = 125 (Cannot assign requested address). Is there an issue
> with my syntax, since I don't see why SIPP shouldn't be able to access
> Kamailio's IP address.
>
> Thankfully
> David
>
>
>
>
> -Original Message-
> From: Rob Day
results.
Best,
Rob
On 2 April 2014 19:18, wrote:
> The SIP server that I am using is Kamailio.
>
>
> -Original Message-----
> From: Rob Day
> To: Munzer,David J
> Cc: sipp-users
> Sent: Wed, Mar 26, 2014 1:14 pm
> Subject: Re: [Sipp-users] Connecting phone sys
ase. I'd
expect to implement some RTP/RTCP support in the next few weeks.
Best,
Rob
On 28 March 2014 14:01, Vijay Goje wrote:
> Hi Rob,
>
> I am looking forward for this.
> Does the jSIPp support rtpstream play?
>
> Regards,
> Vijay.
>
> -Original Message-
t;>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
Hi all,
I've been experimenting recently with a ground-up rewrite of SIPp, to
simplify the code, make it more portable, easier to integrate with
other tools, and faster to develop new features for. I expect to still
maintain the existing C++ version, but the complexity of the code
makes real innov
David,
When you say that you have a phone system running, do you mean that
you have a SIP server (Kamailio/Clearwater/Asterisk) running, or
something else?
If you have a SIP server, it is probably listening on port 5060
(though you can check by running `netstat -lnp`) and you can just give
the IP
/
> message log.
>
>
>
> This is the command I used to initiate the call.
>
> /usr/local/bin/sipp -i 10.ss.ss.ss -s 60661 10.cc.cc.cc -l 5 -r 2 -rp 2s -m
> 5 -sf webex_test.xml
>
>
>
> BR
>
> Brett
>
> From: Brett Harding (AU)
> Sent: Monday, 24 Ma
Demir,
Is it possible that because of the incomplete calls, SIPp has reached
its limit of open calls and stopped creating new ones? You can check
this by looking for a line such as '30 calls (limit 30)' in the
ncurses screen.
If this is the issue, then increasing this limit (with the
command-line
This will leave the call open, allowing SIPp to handle retransmissions
- but means that if the TCP connection closes during that final pause,
the call will fail.
There is a attribute (not very well documented) which can
only go at the end of a scenario, and behaves like except that
it ignores TC
It might be useful to look at
http://sipp.sourceforge.net/doc/reference.html#Structure+of+client+%28UAC+like%29+XML+scenarios
- in particular, I think you can do this by setting 'rrs="true"' on
the received 200 OK, and then using [next_url] for the Request-URI of
the ACK.
Best,
Rob
P.S. I don't t
Hi Massimo,
Are you running SIPp v3.4.1 (which you can get from
https://github.com/SIPp/sipp/releases/tag/v3.4.1)? I improved the
logging of this error in that release - running with that should give
a better indication of what's going on (e.g. why the RTP send is
failing).
Best,
Rob
On 21 March
Is it possible for you to run the test again with -trace_msg and
-trace_err? That should produce log files that give more detailed
information on what's going wrong (for example, any Call-IDs being
reused).
Thanks,
Rob
On 18 March 2014 07:26, Brett Harding (AU)
wrote:
> Hello,
>
>
>
> I have an
Hat 9, and
if not, to unset it on CentOS?
On 11 March 2014 18:50, Rob Day wrote:
> Thanks for reporting this issue - which version of SIPp (and retrieved
> from where) are you trying to compile?
>
> Best,
> Rob
>
> On 11 March 2014 17:44, Steve Edwards wrote:
>> Hi
>&g
Thanks for reporting this issue - which version of SIPp (and retrieved
from where) are you trying to compile?
Best,
Rob
On 11 March 2014 17:44, Steve Edwards wrote:
> Hi
>
> Having compiled and run sipp for some years now under RedHat 9, I'm trying
> to build it under Centos 6.5.
>
> I do make
I've never tried it, but it should be possible by combining a regexp
action to extract the port and the exec action to log it to stdout:
On 21 February 2014 07:27, Alan Martinovic wrote:
> Hi,
>
> I'm doing a simple call simulation with sip
>
> sipp -d 3 -m 1 -i -sf uac.xml
>
ol for
details.
The other is that there is a fix which looks potentially relevant
(https://github.com/SIPp/sipp/commit/86a19680ca64c8ea44569d965dfc46db2c64e01b)
but which didn't quite make it into v3.4.0. So I think taking the
latest git code may well fix your issue.
Best regards,
Rob
On 19 Feb
Thanks - I'll set up a Fedora 20 VM tonight and see if I can reproduce the bug.
On 17 February 2014 20:48, Carlo Carrano
wrote:
> Fedora 20.
>
> Carlo R. Carrano
>
>
>
> Rob Day wrote:
>
>> Thanks for that - what about operating system and version (e.g. Mac
know the results.
>
> Thanks,
>
>
> Carlo R. Carrano
> Product Development Engineer - Research And Development
> Call and Session Control Servers - Dept. NA10090741
> 5420 CTS & SCG Development, Sustaining, and Project Management Team
> Tel: +1-630-713-8911
> OnNET: 287-
Hi Carlo,
Does this reproduce with the latest code from
https://github.com/SIPp/sipp? (You may need to install
autoconf-archive and run './autoreconf -ivf' if it doesn't build
successfully with './configure && make'.)
Assuming it does reproduce, can you let me know what operating system
and versi
Forgot to cc the list on my response.
-- Forwarded message --
From: Rob Day
Date: 10 February 2014 19:39
Subject: Re: [Sipp-users] confused on stable release numbers
To: John Emrich
Hi John,
I moved the SIPp code, bug tracking and downloads from Sourceforge to
Github before
d succeed if AX_HAVE_EPOLL_PWAIT has,
>>> but not the
>>> aclocal.m4:AC_DEFUN([AX_HAVE_EPOLL], [dnl
>>> aclocal.m4:AC_DEFUN([AX_HAVE_EPOLL_PWAIT], [dnl
>>> configure.ac:AX_HAVE_EPOLL([AX_CONFIG_FEATURE_ENABLE(epoll)],
>>>
>>> I'm digging in to see if
tus] Error 2
>
> This is with the latest master that you just committed.
>
>
>
> On Mon, Feb 3, 2014 at 6:43 PM, Rob Day wrote:
>>
>> What command are you running when you get that error? SIPp is built
>> with autoconf/automake, and I've checked in the pre-generated fi
NU Standards-compliant Makefiles
>
>
>
>
>
> On Mon, Feb 3, 2014 at 6:26 PM, Daniel Goepp wrote:
>>
>> Thanks for the follow up. For what it is worth, I just checked out this
>> version, and now get this trying to build it:
>>
>> make: *** No rule to mak
ks for the follow up. For what it is worth, I just checked out this
> version, and now get this trying to build it:
>
> make: *** No rule to make target `src/auth.c', needed by `sipp-auth.o'.
> Stop.
>
>
> On Mon, Feb 3, 2014 at 6:03 PM, Rob Day wrote:
>>
>
Thanks for raising this - it seems to be a side-effect of the move to
epoll (for better performance with high numbers of sockets) in v3.4
(e.g. https://code.google.com/p/dart/issues/detail?id=2775 sees the
same issue). I've just committed a fix to warn rather than error out
in this specific case
(h
Hi Eric,
SIPp doesn't currently support many statistics on RTP packets (though
it's something I might consider adding in the future). Until then, in
your situation, what I'd probably do is to capture all the RTP packets
with Wireshark/tcpdump, then write a small program using Python and
scapy (or
> -skip_rlimit flag fixed my issue and able to run more than 508 concurrent
> calls with rtp.
>
> Regards,
> Vijay.
>
> -Original Message-
> From: Vijay Goje
> Sent: Tuesday, January 28, 2014 6:11 PM
> To: 'Rob Day'
> Cc: sipp-users@lists.s
>
> Tel: +44(0)20 7100 1499
>
> Mob: +44(0)75 1724 7059
>
> Internet communications are not secure and therefore Globility Limited
> does not accept legal responsibility for the contents of this message.
> Any views or opinions presented do not necessarily represent those of
>
I'm not aware of any built-in limit. If you run with "-trace_err" and
"-trace_calldebug", do those logs have any more information on what
happens after 508 calls?
Best,
Rob
On 28 January 2014 22:46, Vijay Goje wrote:
> Hi Rob,
>
>
>
> I am using SIPp v3.4-beta2 with rtp stream.
>
> When I am run
t; Mob: +44(0)75 1724 7059
>
> Internet communications are not secure and therefore Globility Limited
> does not accept legal responsibility for the contents of this message.
> Any views or opinions presented do not necessarily represent those of
> Globility Limited unless otherwise
On 28 January 2014 14:36, Kayode Olajide wrote:
> Hi all,
>
> I am seeing the above error when I run sipp with a xml file that plays a
> message. Sipp crashes after bout 1000 calls have been created.
>
> I am running it with the command sipp -sf uas_custom_loop.xml using the
> latest 3.3
Hi Kay
> On 31 December 2013 14:08, sangdrax8 wrote:
>> I have been testing sipp 3.4 (3.399?) and using the normal
>> distribution pause function. While doing this I have seen that after
>> some time calls begin to get stuck in the pause and never exit that
>> state. I let it run for a week, and eventu
> On 21 January 2014 22:25, wrote:
>> The SIPp 3PCC mode with the sendCmd and recvCmd can be used to exchange
>> out-of-band information between uas and
>> uac if the App under test is a SIP Proxy, where the Call-ID is unmodified
>> and thus known by both uas and uac SIPp processes.
>> But when
Hi,
I think this is probably due to the use of TCP - because TCP is a
connection-oriented protocol, it's common to simply send messages back
over the TCP connection the REGISTER came in on, rather than
constructing a new connection based on the Contact header.
I take it that you need to use TCP r
On 21 January 2014 22:25, wrote:
> The SIPp 3PCC mode with the sendCmd and recvCmd can be used to exchange
> out-of-band information between uas and
> uac if the App under test is a SIP Proxy, where the Call-ID is unmodified and
> thus known by both uas and uac SIPp processes.
> But when testin
Hi Moritz,
The error log you posted says:
2014-01-20 11:17:57:1941390213077.194807: Call-Id: 1-20926@1.2.3.4,
receive timeout on message Test LCR:1 without label to jump to
(ontimeout attribute): aborting call.
2014-01-20 11:17:57:1951390213077.195609: Dead call 1-20926@1.2.3.4
ion from 127.0.0.1 to
> open-ims.test (in run_1.sh script generated by IMS Bench), again because of
> the mentioned reason.
>
> I did No changes to the default scenarios, by the way I've attached them too
> along with manager.xml, ims_bench.xml and report.xml.
>
> Ple
Hi Mohammed,
I don't know a great deal about the IMSBench extensions, I'm afraid,
so I may not be much help - but I can take a look into this.
Could you send the SIPp XML files that IMSBench is using (I don't
think they were attached to your original email), and any information
you can give about
Could you try running with -trace_calldebug? That should log out a bit
more information about pause durations.
My guess is that your pause duration is negative (~0.1% of those
pauses will be more than three standard deviations below the mean,
i.e. <0) and I wouldn't be surprised if the scheduler c
Hi Sangdrax,
As you've found, the situation where you need to register before
receiving a call isn't well-supported in SIPp, particularly not for
connection-oriented transports like TCP. I've just replied to a
separate email thread discussing how out-of-call scenarios can help
here (and what their
ff and there are example scripts
at https://github.com/rkday/quaff-examples and
https://github.com/Metaswitch/clearwater-live-test/tree/master/lib/tests.
Best,
Rob Day
(SIPp maintainer)
On 15 January 2014 19:11, wrote:
> Dragos, Thanks for your response.
>
> I did in fact, modified U
Hi Richard,
It looks like that warning is there in the 3.2 docs as well, which
means it dates from before I started working on SIPp. So I'm afraid I
don't know the reason behind it, and haven't done any RTP-over-IPv6
testing which would confirm it. If you try it, do let me know how you
get on so I
Richard,
I just wanted to let you know that I've now acquired a Mac OS X 10.7
system and can reproduce this odd terminal behaviour, so will try and
get it fixed for version 3.5.
Best,
Rob
On 27 October 2013 14:38, Rob Day wrote:
> Yes, I agree that it's not a very clean solution -
On 24/12/13 20:21, Divin John (dijohn) wrote:
HI all,
I am trying to achieve the following:
UACUAS
Invite -->
<--- TRYING
<--- 180/183
Update --->
<-- 200 OK for UPDATE
<--- 200 OK for INVITE
ACK >
SIPP doesn't even execute this. Am I missing something obvious? If I
move the UPDATE to be
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