Re: [Sipp-users] What is dead call?
If a call has completed, then SIPp maintains a small record of the call for a little bit. This way any messages related to that call; after the call has already been closed can be cataloged as dead call messages. Without keeping this record, the messages for a completed call are counted as out-of-call messages on the UAC. On the UAS, a new call is automatically created and fails because the message is, most likely, unexpected. Charles Manish Sapariya man...@gslab.com wrote on 04/23/2010 01:26:44: Manish Sapariya man...@gslab.com 04/23/2010 01:26 To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] What is dead call? I tried to google and even looked at the source. What I understood, is that 'a deadcall, is a call for which there was not responses until deadwait time'. Can somebody please confirm. If my understanding is correct, then how sipp responds to or accounts for the reply received after deadwait period. Thanks and Regards, Manish -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users-- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] label, next and ontimeout broken in 3.1
I think the best answer would be to strdup the return from xp_get_value, as the named labels are far friendlier when writing any complex scenario that uses more than a handful of labels. Charles Peter Higginson plh...@hotmail.c omTo sumeet_bhard...@persistent.co.in, 01/13/2010 16:08 sipp_users sipp-users@lists.sourceforge.net cc Subject [Sipp-users] label, next and ontimeout broken in 3.1 The C routine xp_get_value returns a pointer to a static buffer from which a value is normally extracted, or in a few cases a string is copied to a new buffer. The new label code takes this pointer and uses it as the str part of an int_str_map. So it's just junk - it points to whatever last used the xp_get_value routine and it's a serious bug. It impacts all labels so I have changed the title of this message. Unless there is some magic way to get map to copy the elements (I don't know one), I can see a hard way (make an explicit copy of the strings) or an easy way (restrict the labels to integers) to fix this. Changing both the str_int_map and the int_str_map to int_int_map would be fairly simple to do. It would give you arbitrary integers as labels and keep the error checking advantage of the new code and allow large numbers of labels. Integers are the only things documented so I doubt many scenarios have non-numeric labels. The next step is upto the maintainers. The only thing I can suggest to Sumeet is to use version 3.0 which allows labels 1-99 only but has the old working code. Peter From: plh...@hotmail.com To: sumeet_bhard...@persistent.co.in; sipp-users@lists.sourceforge.net Date: Wed, 13 Jan 2010 18:11:19 + Subject: Re: [Sipp-users] FW: Help- ontimeout is not working Being snowed-in in Southern England (it is just so infrequent that we do not prepare for it) I am having a look at this in a bit more depth. The report so far is: It looks like the forward reference resolver is broken. So backward references work and all forward references go to the last defined label. I have a trap in scenario::apply_labels which shows this happening. -- The history is that when I wrote the label and next stuff I avoided having a resolver pass by limiting the labels to 1-n. (n was 9 initially - now 99.) My logic was that scenarios should be simple with few labels and the array required was smaller than the code for a resolver pass. Someone later wrote a resolver pass to get more flexibility and error checking - I'm fine with this but it means I'm now looking at code that is new to me. -- If you simplify your scenario to only use one label or only use backward references then it will work. (However I suspect that is hard to do.) Peter Higginson From: sumeet_bhard...@persistent.co.in To: sipp-users@lists.sourceforge.net Date: Tue, 12 Jan 2010 15:21:44 +0530 Subject: [Sipp-users] FW: Help- ontimeout is not working Please reply if anyone knows about this. Thanks -Sumeet From: Sumeet Bhardwaj Sent: Monday, January 11, 2010 6:05 PM To: 'Peter Higginson'; dushyant.dha...@rancoretech.com Cc: sipp_users Subject: RE: [Sipp-users] Help- ontimeout is not working Hello All, I am using sip 3.1 version. Only using uac xml And command for the same is : sipp -sf Sample.xml ip:port -inf input csv file -r 1 -rp 1s -nr -trace_msg Thanks -Sumeet From: Peter Higginson [mailto:plh...@hotmail.com] Sent: Monday, January 11, 2010 5:00 PM To: dushyant.dha...@rancoretech.com; Sumeet Bhardwaj Cc: sipp_users Subject: RE: [Sipp-users] Help- ontimeout is not working Also the exact version you were using What you have to watch is what state you are in. (When all else fails put a trace on the state.) I'm retired now and not
Re: [Sipp-users] Query- watchdog Functionality
You can increase the timeout, but if you are getting watch dog timeouts, it means that you do not have enough resources on the machine to generate the amount of traffic that you are asking for. Charles Sumeet Bhardwaj sumeet_bhard...@persistent.co.in 09/30/2009 06:21 AM To sipp-users@lists.sourceforge.net sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Query- watchdog Functionality Hello All, I am running SIPp and getting failure due to watchdog functionality. Does anyone know what is this functionality and how to disable this? Error : sipp: The following events occured: 2009-09-30 09:03:29:1801254301409.180100: The minor watchdog timer 500ms has been tripped (564), 120 trips remaining.. Thanks -Sumeet DISCLAIMER == This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails. -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Kill key?
Try Q. Charles Mike Ayers mike_ay...@tvworks.com 05/07/2009 08:05 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Kill key? I'm using SIPP 3.x (2009-01-21 unstable). I've found that 2 qs don't make a kill, unlike 2.x. I'm trying to kill a server by remote control - is there a keystroke to put the server out of my misery? This also applies to interactive mode, yes? Thanks, Mike -- The NEW KODAK i700 Series Scanners deliver under ANY circumstances! Your production scanning environment may not be a perfect world - but thanks to Kodak, there's a perfect scanner to get the job done! With the NEW KODAK i700 Series Scanner you'll get full speed at 300 dpi even with all image processing features enabled. http://p.sf.net/sfu/kodak-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- The NEW KODAK i700 Series Scanners deliver under ANY circumstances! Your production scanning environment may not be a perfect world - but thanks to Kodak, there's a perfect scanner to get the job done! With the NEW KODAK i700 Series Scanner you'll get full speed at 300 dpi even with all image processing features enabled. http://p.sf.net/sfu/kodak-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How can i config SIPp send response through VIA header?
You can use regular expression matching and the setdest action. Charles hui cheng avantasia2...@gmail.com 04/07/2009 04:58 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] How can i config SIPp send response through VIA header? hi all How cant set SIPp send response through the ipaddress get from the first VIA header? I guess SIPp send on ther same socket which request sends into. In some situation , it will have some impact. Thanks -- This SF.net email is sponsored by: High Quality Requirements in a Collaborative Environment. Download a free trial of Rational Requirements Composer Now! http://p.sf.net/sfu/www-ibm-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: High Quality Requirements in a Collaborative Environment. Download a free trial of Rational Requirements Composer Now! http://p.sf.net/sfu/www-ibm-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Multiple rtd in recv
The XML parser would ideally throw an error, but it doesn't. To get this behavior you'll need to use some gotos and nops. Charles Artem Naluzhnyy t...@nhamon.com.ua 03/30/2009 09:05 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Multiple rtd in recv Hi, Only first rtd parameter is processed in following code: recv response=180 optional=true rtd=PDD_180 rtd=PDD/recv Bug/feature? -- Artem Naluzhnyy -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Storing the result of an external command into an internal variable
Not without modifying the SIPp code.
Charles
Madiha Shahid madis1...@gmail.com
03/26/2009 08:50 AM
To
sipp-users@lists.sourceforge.net
cc
Subject
[Sipp-users] Storing the result of an external command into an internal
variable
Hi,
I want to store the result of an external command into an internal
variable. Is there a way to do that?
Example:
exec command=echo `sndfile-info [$filename] | grep Duration | awk
'{print $3}'`/
Is there a way we can extract the value returned by the above command into
an internal variable so that I can manipulate it internally in the xml
script?
Kindly respond.
Regards,
Madiha Taha
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Re: [Sipp-users] A more robust SIPp
Unfortunately, SIPp uses a built-in XML parser using strstr. Charles Kirwan, David (David) dkir...@avaya.com 03/13/2009 06:29 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] A more robust SIPp Hi, While writting a SIPp script, I omitted a forward slash. The script ran, but didn't send the ACK, which was a few lines below the line of XML that was missing the forward slash. I'm sure SIPp has a fully fledged XML parser onboard, probably IBM's xerces parser, so why can't SIPp tell you the XML is or isn't well-formed and valid before trying to run the script? Best regards, David -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] -trace_rtt
I think this could be an old version which used cout, and hence scientific notation for large numbers. What version is this from? More recent versions use printf, which should do regular decimal numbers. Charles 鄰家老王 wgh65...@hotmail.com 02/16/2009 11:15 PM To Sipp-users@lists.sourceforge.net cc Subject [Sipp-users] -trace_rtt Hi I have a question , please help me, i run sipp use -trace_rtt, but outputfile _3316_rtt.csv dispaly follow, i don't know , happen rtt Xe+06 reason? - 4.28105e+06;0;1 4.28622e+06;0;1 4.29141e+06;0;1 4.29656e+06;4.29497e+06;1 4.30175e+06;4.29497e+06;1 4.30692e+06;4.29497e+06;1 4.31211e+06;4.29497e+06;1 Regards, dickson -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Open Source Business Conference (OSBC), March 24-25, 2009, San Francisco, CA -OSBC tackles the biggest issue in open source: Open Sourcing the Enterprise -Strategies to boost innovation and cut costs with open source participation -Receive a $600 discount off the registration fee with the source code: SFAD http://p.sf.net/sfu/XcvMzF8H ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to extract the second the second string from the VIA: header
You need to have parenthesis to capture just the part after the comma.
Something like.
ereg regexp=,(foo) assign_to=dummy,foovalue /
Charles
Pratap Nath pratapn...@gmail.com
02/02/2009 08:43 AM
To
sipp-users@lists.sourceforge.net
cc
Subject
[Sipp-users] How to extract the second the second string from the VIA:
header
HI All,
I was simulating a conference scenario with Sipp. The Switch sends me a
update message with following Information.
UPDATE sip:5...@148.147.171.54:64478;transport=tcp SIP/2.0
Call-ID: 15_374cb53-41c226674cf7e31...@148.147.171.54
CSeq: 2 UPDATE
From: sip:5...@avaya.com;tag=8038c7bbbebdd17a3497b5ad100
To: sip:5...@avaya.com;tag=-2cee5336498317ae-c_F148.147.171.54
Via: SIP/2.0/TCP
148.147.171.28:5060;branch=z9hG4bK83138303530393F5334001b.0,SIP/2.0/TLS
148.147.171.28:6001;psrrposn=1;received=148.147.171.28;branch=z9hG4bK809229ebbebdd1893497b5ad100
Content-Length: 0
Contact: CONFERENCE 2 sip:148.147.171.28:6001;transport=tls;isfocus
Max-Forwards: 69
User-Agent: Avaya CM/R015x.01.2.416.4
Allow:
INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH
Supported: 100rel,timer,replaces,join,histinfo
Min-SE: 180
Session-Expires: 180;refresher=uac
Record-Route: sip:148.147.171.28:5060;transport=tcp;lr
Correct response .
SIP/2.0 200 OK
From: sip:5...@avaya.com;tag=8038c7bbbebdd17a3497b5ad100
To: sip:5...@avaya.com;tag=-2cee5336498317ae-c_F148.147.171.54
Call-ID: 15_374cb53-41c226674cf7e31...@148.147.171.54
CSeq: 2 UPDATE
Via: sip/2.0/tcp
148.147.171.28:5060;branch=z9hG4bK83138303530393F5334001b.0
Via: SIP/2.0/TLS
148.147.171.28:6001;psrrposn=1;received=148.147.171.28;branch=z9hG4bK809229ebbebdd1893497b5ad100
Record-Route: sip:148.147.171.28:5060;transport=tcp;lr
User-Agent: Avaya one-X Deskphone
Content-Length: 0
As Sipp is receiving the UPDATE successfully but While sending the 200OK
for UPDATE the SES does not process the 200OK as it wants that the 200OK
which sipp sends should contains
two separate via headers .
In my case i send 200OK which contains the following dump
SIP/2.0 200 OK
From: sip:5...@avaya.com;tag=8038c7bbbebdd17a3497b5ad100
To: sip:5...@avaya.com;tag=-2cee5336498317ae-c_F148.147.171.54
Call-ID: 15_374cb53-41c226674cf7e31...@148.147.171.54
CSeq: 2 UPDATE
Via: sip/2.0/tcp
148.147.171.28:5060;branch=z9hG4bK83138303530393F5334001b.0,SIP/2.0/TLS
148.147.171.28:6001;psrrposn=1;received=148.147.171.28;branch=z9hG4bK809229ebbebdd1893497b5ad100
Record-Route: sip:148.147.171.28:5060;transport=tcp;lr
User-Agent: Avaya one-X Deskphone
Content-Length: 0
My final approach which i used to send was followed the regular expression
path .
I have used two regular expression commands to separate two strings from
the Via : header
The Regular expression
=
1. ereg regexp= .[^,]+ search_in=hdr header=Via: assign_to =2/
2. ereg regexp=\,([A-Z]{3}/2\.[0-9]/[A-Z]{3}) search_in=hdr
header=Via: assign_to =3/
the variable $2 gets sip/2.0/tcp
148.147.171.28:5060;branch=z9hG4bK83138303530393F5334001b.0
and the variable $3 gets ,SIP/2.0/TLS
148.147.171.28:6001;psrrposn=1;received=148.147.171.28;branch=z9hG4bK809229ebbebdd1893497b5ad100
The problem is i was unable to erase the , (comma) from the string
which is stored in the $3 .
the sample of 200OK is shown below after applying the regular expression :
SIP/2.0 200 OK
From: sip:5...@avaya.com;tag=8038c7bbbebdd17a3497b5ad100
To: sip:5...@avaya.com;tag=-2cee5336498317ae-c_F148.147.171.54
Call-ID: 15_374cb53-41c226674cf7e31...@148.147.171.54
CSeq: 2 UPDATE
Via: sip/2.0/tcp
148.147.171.28:5060;branch=z9hG4bK83138303530393F5334001b.0
Via: ,SIP/2.0/TLS
Via:148.147.171.28:6001;psrrposn=1;received=148.147.171.28;branch=z9hG4bK809229ebbebdd1893497b5ad100
Record-Route: sip:148.147.171.28:5060;transport=tcp;lr
User-Agent: Avaya one-X Deskphone
Content-Length: 0
If somebody has any idea how to get the second string without , .
Please advice .
Thanks
Pratap Nath
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Re: [Sipp-users] Segmentation fault provoked by stack corruption while using big ResponseTimeRepartition or CallLengthRepartition values exceeding 1024 bytes in length
Artur, Thank you for the bug report.Subversion Revision 570 should correct this issue permanently. Charles Artur silveira da Cunha artur.silveira_da_cu...@alcatel-lucent.fr wrote on 01/26/2009 07:04:30 PM: Hi, sipp crashes with Segmentation fault error when in the scenary the ResponseTimeRepartition or CallLengthRepartition parameters are filled with values that bypasses the 1024 bytes in length. I search for the error and found that in stat.cpp and stat.hpp the buffers are sized to 1024 bytes and that the buffers uses strcat or sprintf instructions that create corruption stack situations when the value to be copied exceeds the receiving buffer length.. I made the following correction in stat.hpp to increase the buffer values from 1024 to 2048 and now sipp no more crashes. I know that this correction is not final and that it pushes the problem more far when the parameters values will exceeds 2048 bytes in length. Corrected lines: Before #define MAX_REPARTITION_HEADER_LENGTH 1024 #define MAX_REPARTITION_INFO_LENGTH 1024 #define MAX_CHAR_BUFFER_SIZE 1024 After #define MAX_REPARTITION_HEADER_LENGTH 2048 #define MAX_REPARTITION_INFO_LENGTH 2048 #define MAX_CHAR_BUFFER_SIZE 2048 Note: We are using an old 2007 sipp version with which we don't have this problem, it's now that we try to use new sipp versions that we find this problem. I compare the source code and found that many changes has been made in stat.hpp and stat.cpp source code. Regards Artur ?xml version=1.0 encoding=ISO-8859-1 ? !DOCTYPE scenario SYSTEM sipp.dtd !-- This program is free software; you can redistribute it and/or -- !-- modify it under the terms of the GNU General Public License as -- !-- published by the Free Software Foundation; either version 2 of the -- !-- License, or (at your option) any later version. -- !-- -- !-- This program is distributed in the hope that it will be useful, -- !-- but WITHOUT ANY WARRANTY; without even the implied warranty of -- !-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -- !-- GNU General Public License for more details. -- !-- -- !-- You should have received a copy of the GNU General Public License -- !-- along with this program; if not, write to the -- !-- Free Software Foundation, Inc., -- !-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -- !-- -- !-- Sipp default 'uac' scenario. -- !-- -- scenario name=Basic Sipstone UAC !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword. -- send retrans=500 ![CDATA[ INVITE sip:[servi...@localhost:5060 SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp sip:s...@[local_ip]:[local_port];tag=[pid] SIPpTag00[call_number] To: sut sip:[servi...@[remote_ip]:[remote_port] Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:s...@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]] /send recv response=480 optional=true /recv !-- definition of the response time repartition table (unit is ms) -- !-- definition of the call length repartition table (unit is ms) -- !-- added by sippStatsMaker - definition of the response time repartition table (unit is ms) -- ResponseTimeRepartition value = 1,2,3,4,5,6,7,8,9,10,11,12,13, 14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,35, 36,37,38,39,40,41,42,43,44,45,46,47,48,49,50,51,52,53,54,55,56,57, 58,59,60,61,62,63,64,65,66,67,68,69,70,71,72,73,74,75,76,77,78,79, 80,81,82,83,84,85,86,87,88,89,90,91,92,93,94,95,96,97,98,99,100,101, 102,103,104,105,106,107,108,109,110,111,112,113,114,115,116,117,118, 119,120,121,122,123,124,125,126,127,128,129,130,131,132,133,134,135, 136,137,138,139,140,141,142,143,144,145,146,147,148,149,150,151,152, 153,154,155,156,157,158,159,160,161,162,163,164,165,166,167,168,169, 170,171,172,173,174,175,176,177,178,179,180,181,182,183,184,185,186, 187,188,189,190,191,192,193,194,195,196,197,198,199,200,201,202,203, 204,205,206,207,208,209,210,211,212,213,214,215,216,217,218,219,220, 221,222,223,224,225,226,227,228,229,230,231,232,233,234,235,236,237, 238,239,240,241,242,243,244,245,246,247,248,249,250,251,252,253,254, 255,256,257,258,259,260,261,262,263,264,265,266,267,268,269,270,271, 272,273,274,275,276,277,278,279,280,281,282,283,284,285,286,287,288, 289,290,291,292,293,294,295,296,297,298,299,300/ !-- added by sippStatsMaker - definition of the call length repartition table (unit is ms) --
Re: [Sipp-users] Can sipp call rate and call parameter be controlled by external script/program
You can tell SIPp to change its rate using the control socket. Can I tell sipp, not to make next call, until I am ready again? You can set the rate to zero. You can probably change the call parameters using extended 3PCC. If you establish a socket to a 3PCC element and setup various parameters (look at the replace and insert actions so that you can update the in-memory representation of the CSV file); you can then pass those parameters to another instance of SIPp that will generate the calls (or you might even be able to generate the calls from that instance). I will study 3PCC in more details, but can I pass the parameter from my external script to 3PCC? Only if your external script speaks 3PCC. It is relatively simple. You open a TCP connection to the 3PCC port, and send messages terminated by \27 (i.e. the character 27. The messages can contain anything you want. The messages are SIP-like, and are something like the following: Call-ID: Foo From: class Any-Random-Header: Value Body text\27 Where Foo is the call ID you select, From is the names in slave.cfg, and Any-Random-Header and Body text are the infomration you want to put in. SIPp internally adds the \27, but if you need an external script you'll need to do it yourself. If you do modify the injection file, I would suggest using something like a MySQL database as a backing store and querying the database. I believe this is alternative to my socket based approach. However if 3PCC approach works, I may not need to need this. In any case, could you please point me to the files, that I should start looking at? message.cpp defines the keywords (the SendingMessage class parses the XML into a structure); call.cpp (create_sending_message) interprets the structure created by message.cpp; and infile.cpp handles the file keywords. Charles -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Can sipp call rate and call parameter be controlled by external script/program
You can tell SIPp to change its rate using the control socket. You can probably change the call parameters using extended 3PCC. If you establish a socket to a 3PCC element and setup various parameters (look at the replace and insert actions so that you can update the in-memory representation of the CSV file); you can then pass those parameters to another instance of SIPp that will generate the calls (or you might even be able to generate the calls from that instance). If you do modify the injection file, I would suggest using something like a MySQL database as a backing store and querying the database. You really really want to avoid anything that will block in SIPp code (because it is single threading blocking in one place will delay all traffic processing). Charles Manish Sapariya man...@gslab.com 01/22/2009 11:05 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Can sipp call rate and call parameter be controlled by external script/program Hi All, Little background: == We have openser/asterisk/openfire based collabration system. XMPP is used extensively as control channel amongst the client. For establishing the calls, the clients first register using XMPP and then request SIP parameters using XMPP control message. The sip stack then uses these extra headers that are returned in XMPP response to establish sip/audio calls. What I need: I have XMPP client script, which can register as multiple users and get the extra SIP header information. My question is, can I ask sipp to initiate call whenever my xmpp user is ready with extra sip params, and make sipp call using these params and at the rate at which xmpp script is asking sipp to make call? Any hints would be appreciated. If not possible in in stock sipp, any hints regarding what area of code might need changes. One area of sipp (though I have not looked at code in great details), is the part where CSV based parameter handling is done. What I think should be possible is to replace CSV reading part of code with socket based reading. If I can open a socket for reading the next call parameters and block until these parameters are not receivede, my external script should be able to send call params by sending data to this socket. Does this sound workable? Any other alternative? Thanks and Regards, Manish -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Calls limit
By default, SIPp is closed loop and will not allow more than 3 * call duration (with a minimum of one second) calls to be outstanding. Specify -l 0 to get open loop behavior, which removes this limit. Charles Evgeny Miloslavsky emiloslav...@juniper.net 01/21/2009 04:46 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Calls limit Hi While trying to create traffic of 30 REGISTER/sec during 5 seconds (that is 150 REGISTERS totally), I saw that only 90 REGISTERs are created and there is a following output at scenario screen: 90 calls (limit 90) What is this limit? Is this value changeable? And if it does how do I change it. Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9717320/2355 -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Random calls in SIPp
There is no way to do it without modifying the SIPp source code or using 3PCC. To modify the source code, you should look at opentask.cpp. For 3PCC, you need to have a master controller scenario that pauses a random amount, then kicks another scenario to do the actual call. Charles Katarina Bogdan katarin...@net.hr 01/20/2009 08:04 AM Please respond to katarin...@net.hr To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Random calls in SIPp Can I get multiple calls in SIPp to appear random I'm using this command ... -r 2 -rp 10s -m 50 ... and of course it works, but It's too uniform. Thanks agein! -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Can SIPp run two scenarios simultaneously?
You can't do this without using the same Call-ID or modifying the source. Charles Katarina Bogdan katarin...@net.hr 01/20/2009 07:42 AM Please respond to katarin...@net.hr To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Can SIPp run two scenarios simultaneously? This is the problem: I have to establish Sip session, then my client sends HTTP POST to change his servce profile (this issue is solved automatically) and then SERVER sends him UPDATE request with new profile and there lies the problem. How to do that if SIPp does not alowe sending and receiveing request in one scenario? If I break that in two scenarios, how to run them simultaneously with the same TCP connection? Thanks for help CLIENT SERVER INVITE 100--- 180--- 183--- PRACK- 200 OK(PRACK)- UPDATE 200 OK(UPDATE) 200 OK(INVITE) ACK--- UPDATE 200 OK(UPDATE) ACK--- HTTP POST- UPDATE 200 OK(UPDATE) UPDATE 200 OK(UPDATE) -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] size of xml scenarios
Yes. You may be able to change this line in xp_parser.c to get a larger limit. Charles #define XP_MAX_FILE_LEN 65536 Katarina Bogdan katarin...@net.hr 01/19/2009 10:23 AM Please respond to katarin...@net.hr To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] size of xml scenarios Is there a limit in size of xml scenarios. I have one that´s 90KB, and when I try to run it with SIPp I´m getting this error Unable to load or parse xml scenario -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Can client receve request and send reply
The scenario can either send or receive requests, but can not be both a client (initiate brand new scenarios) and server (accept brand new scenarios). A scenario is determined by call-id. Charles Katarina Bogdan katarin...@net.hr 01/19/2009 10:05 AM Please respond to katarin...@net.hr To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Can client receve request and send reply I have a client like scenario that goes like this: INVITE 100--- 180--- 183--- PRACK- 200 OK(PRACK)- UPDATE 200 OK(UPDATE) 200 OK(INVITE) ACK--- UPDATE 200 OK(UPDATE) ACK--- UPDATE 200 OK(UPDATE) UPDATE 200 OK(UPDATE) And a server scenario compatible with this one writen in xml. When I try to launch this in sipp (sipp -sf server.xml -t t1) I´m getting this error : Unable to load or parse xml scenario file There are no errors in scenario or in commands. Please help! Is it even posible that one scenario have a role of both, client and a server, I´m refering to last 4 messages -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] how to timeout on receive of a message
You can use an ontimeout attribute on your receive for the ACK. The usage should be in the reference document. Charles Amir Abdollahi aabdol...@yahoo.com 01/19/2009 09:09 AM Please respond to aabdol...@yahoo.com To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] how to timeout on receive of a message Hi, I have a UAS.XML script (server) that sometimes does not recieve an ACK message because the phone that acts as client dies. I was told that this will leave the phone call as setup and after a while I am not able to make calls to the server any more...Is this true? Second questions: How can I setup my UAS.XML script to timeout and send a BYE message if the phone hangs and leave the call up? If anyone has a UAS example please send it to me. Thanks Amir -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Sipp scenario that first send a REGISTER and then wait for an INVITE
There is no way to do it in the same scenario without the patch. You can create two scenarios; and if needed tie them together with 3PCC. Charles Ulrik Svensson ulrik.svens...@ericsson.com 01/12/2009 12:16 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Sipp scenario that first send a REGISTER and then wait for an INVITE Hi! I'm trying to build a sipp scenario that first send a REGISTER and then wait for an INVITE, but the INVITE message is discarded with the error message: Discarding message which can't be mapped to a known SIPp call It seems that Raphael Benedet reported the same problem on the mailinglist 2006-09-12 and Olivier Jacques answered that there is a pre-post scenario patch which is targeted to solve this problem. The patch is listed on http://sipp.sourceforge.net/wiki/index.php/Patches. Is anybody working on integrating the patch with the latest sipp release? Or is there a reason for not integrating the patch? Is there any other way that I can send a REGISTER and then wait for an INVITE in the same sipp scenario, without the patch? Thanks in advance! /Ulrik -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Incorrect default behavior in case of SUBSCRIBE-NOTIFY flow
Use -default_behaviors all,-bye. Charles Evgeny Miloslavsky emiloslav...@juniper.net 01/11/2009 02:09 AM To Charles P Wright/Watson/i...@ibmus, sipp-users@lists.sourceforge.net cc Subject Incorrect default behavior in case of SUBSCRIBE-NOTIFY flow Hi While running SUBSCRIBE-NOTIFY flow using SIPp-3.0 I saw that in case SIPp-UAC receives unexpected response on SUBSCRIBE request (in my case it was 403 Forbidden) it sends BYE request to terminate the flow. As far as I understand this behavior is incorrect even if ?nd option fixes the problem. Any ideas? Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 -- Check out the new SourceForge.net Marketplace. It is the best place to buy or sell services for just about anything Open Source. http://p.sf.net/sfu/Xq1LFB ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] rrs=true does not work
The rrs=true should be independent of next. Did you try a SIPp -trace_msg to see what it thinks it is sending? Charles Andreas Winkelbauer andreas.winkelba...@gmx.at wrote on 01/08/2009 11:33:27 PM: Hi, I am currently testing a SIPp scenario and I am experiencing problems with rrs=true and next=... when receiving optional responses. You can find the relevant parts of the scenario below. In my scenario I want to consider some exceptional cases, for example a 480 response to an INVITE request. After receiving such a response I want to end the SIP dialog in a proper way, in this case by sending an ACK message. Now the problem is that this ACK message (at label #12) is never sent (it is skipped without any error message!) as soon as I use either [next_url] or [routes]. Unfortunately I have to use both keywords, since all SIP messages have to be routed via a proxy. The other ACK message in the scenario (at label #9) also uses the keywords [next_url] and [routes], but it is processed and sent flawlessly. Could somebody tell me why the ACK message at label #12 is not sent at all and there is no error message? (Yes, the message is not sent for sure, I traced the SIP messages using Wireshark.) It seems to me as if SIPp is unable to process [next_url] and [routes] if rrs=true is used in conjunction with next=... when receiving an optional response. Is this behavior by design or is it a bug? Any suggestions how I could properly end the SIP dialog in this case? Bye, Andreas Winkelbauer ?xml version=1.0 encoding=ISO-8859-1 ? scenario name=swkn sipp scenario label id=5 / send retrans=500 ![CDATA[ INVITE sip:[field5] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field3] sip:[fiel...@[field0];tag=[call_number] To: [field4] sip:[field5] Contact: sip:[fiel...@[local_ip]:[local_port] Call-ID: [call_id] CSeq: [cseq] INVITE Max-Forwards: 70 User-Agent: [field6] Content-Type: application/sdp Content-Length: [len] ... SDP ... ]] /send !-- consider exceptional cases -- ... recv response=480 optional=true next=12 rrs=true action exec command=echo Error: ... sipp.log / /action /recv ... !-- receive 100 trying (optional) -- label id=6 / recv response=100 optional=true rrs=true /recv !-- receive 180 ringing (optional) -- label id=7 / recv response=180 optional=true rrs=true /recv !-- receive 200 OK -- label id=8 / recv response=200 rrs=true /recv !-- send ACK -- label id=9 / send ![CDATA[ ACK [next_url] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field3] sip:[fiel...@[field0];tag=[call_number] To: [field4] sip:[field5][peer_tag_param] Contact: sip:[fiel...@[local_ip]:[local_port] [routes] Call-ID: [call_id] CSeq: [cseq] ACK Max-Forwards: 70 User-Agent: [field6] Content-Length: 0 ]] /send ... play pcap files ... !-- send BYE -- label id=10 / send retrans=500 ![CDATA[ BYE [next_url] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field3] sip:[fiel...@[field0];tag=[call_number] To: [field4] sip:[field5][peer_tag_param] Contact: sip:[fiel...@[local_ip]:[local_port] [routes] Call-ID: [call_id] CSeq: [cseq] BYE Max-Forwards: 70 User-Agent: [field6] Content-Length: 0 ]] /send !-- receive 200 OK -- label id=11 / recv response=200 next=13 /recv !-- send ACK -- !-- ERROR: this message is never sent! -- label id=12 / send ![CDATA[ ACK [next_url] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field3] sip:[fiel...@[field0];tag=[call_number] To: [field4] sip:[field5][peer_tag_param] Contact: sip:[fiel...@[local_ip]:[local_port] [routes] Call-ID: [call_id] CSeq: [cseq] ACK Max-Forwards: 70 User-Agent: [field6] Content-Length: 0 ]] /send label id=13 / /scenario -- Check out the new SourceForge.net Marketplace. It is the best place to buy or sell services for just about anything Open Source. http://p.sf.net/sfu/Xq1LFB ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Check out the new SourceForge.net Marketplace. It is the best place to buy or sell services for just about anything Open Source. http://p.sf.net/sfu/Xq1LFB ___ Sipp-users mailing
Re: [Sipp-users] Injecting values/variables
Andreas Winkelbauer andreas.winkelba...@gmx.at wrote on 01/09/2009 08:10:15 PM: 1) When using pcap play I would like to inject the path to the pcap file using a [fieldN] keyword or a variable [$n]. So, instead of something like exec play_pcap_audio=/path/to/file.pcap / I would like to have exec play_pcap_audio=[fieldN]/file.pcap / or exec play_pcap_audio=[$n]/file.pcap / I tried this in several ways but it never worked out, so I guess this is just not possible at the moment. Is this correct? Yes, it is not currently possible. I was going to say it would be easy to implement, but the problem here is that the pcap file is loaded and processed on scenario startup; not once per call. It is possible to change this, but it would need some sort of caching to be efficient in the common case of a hardcoded file [also to deal with concurrent calls]. If someone is motivated to do this, I can give pointers. 2) I would like to use meaningful variable names in my scenario. So, for example, instead of writing [field2] I would like to use something like [$auth_string]. To achieve this I was using an action like assignstr assign_to=auth_string value=[field2] / where [field2] is equal to [authentication username=foo password=bar]. But when doing so I got an error message saying Authentication keyword without dialog_authentication!.. This does not happen in cases where [fieldN] contains something different (e.g. normal text). This seems to be a bug, right? The assignstr will try to evaluate what it is substituting, but it can't handle the authentication at this point; because it needs to have the challenge sent. I suggest that instead of embedding the whole auth string, you do something like having this in your injection file, then: SEQUENTIAL field0,field1,foo,bar In your actions: assignstr assign_to=auth_user value=[field2] / assignstr assign_to=auth_pass value=[field3] / And then in the message: [authentication username=$auth_user password=$auth_pass] Charles -- Check out the new SourceForge.net Marketplace. It is the best place to buy or sell services for just about anything Open Source. http://p.sf.net/sfu/Xq1LFB ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] -stat-delimiter parameter
Evegeny, It changes the string ; between fields to a string of your choosing. In Europe, Excel uses ; to differentiate fields, but in the US it uses ,. So we always use -stat_delimiter , (meaning -stat_delimiter [space] [comma]) which separates the fields using a comma instead of a semi-colon. This makes it easier to load the files into our spreadsheet program. You can, however, choose any arbitrary string. Charles Evgeny Miloslavsky emiloslav...@juniper.net wrote on 01/01/2009 09:41:59 AM: Hi All! What is -stat-delimiter parameter and why do we need it? As SIPp help explains it sets the delimiter for the statistics file but I don?t understand delimeter of what and what are the possible delimitations and values Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to use the [userid] keyword when running with -users option
The 2.0 series is very old, and doesn't support lots of new features like this keyword. Charles aleksander.morg...@gmail.com wrote on 12/23/2008 03:35:39 PM: The one shipped in latest Ubuntu GNU/Linux: ii sip-tester2.0.1-1.2 a performance testing tool for the SIP proto Anyway, I am now skipping this issue injecting a CSV file with a userid column, which I guess is the way to go -Aleksander On Tue, Dec 23, 2008 at 9:14 PM, Charles P Wright cpwri...@us.ibm.com wrote: What version of SIPp are you using? Charles Aleksander Morgado sipp-us...@aleksander.es 12/23/2008 02:59 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] How to use the [userid] keyword when running with -users option Hi all, I would like to use the -users N option to always have N calls being run, and also simulate different users at the same time, so I added the [userid] keyword as in this example: send ![CDATA[ REGISTER sip:alekstest.com SIP/2.0 Route: sip:192.168.2.7;lr;transport=UDP Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: User [userid] sip:123456[user...@alekstest.com;tag=[call_number] To: User [userid] sip:123456[user...@alekstest.com Call-ID: [call_id] CSeq: 1 REGISTER Contact: sip:123456[user...@[local_ip];transport=UDP;expires=3600 Supported: sec-agree Expires: 3600 Content-Length: 0 ]] /send But I get the following error: 2008-12-23 20:50:50: Unsupported keyword 'userid' in xml scenario file. I run sipp as follows: $ sipp -sf scenario.xml -r 1 -rp 1000 -users 10 192.168.2.7 Needless to say, I am quite new to sipp, so probably I am doing something really wrong... Any hint? Thanks in advance, -Aleksander -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Help regarding csv file
You can't use keywords in the regexp action. Charles Sumeet Bhardwaj sumeet_bhard...@persistent.co.in wrote on 12/19/2008 05:22:16 AM: Hi all, I am trying to input value from CSV into the scenario file using ? ?inf cvs file name ? command. But SIPp is not extracting value from csv file. Below are the files and command: CVS file: 1. validate.csv SEQUENTIAL 5111313001 Scenario file: 2. SIPp server file: terminator_4GEE_B2BUA_CR1_IAM_0001.xml recv request=INVITE optional=false action ereg regexp=.* search_in=hdr header=CSeq: check_it=true assign_to=2 / ereg regexp=.* search_in=hdr header=Via: check_it=true assign_to=1 / ereg regexp=[field0 line=1] search_in=hdr header=P-Asserted- Identity: check_it=true assign_to=5 / /action /recv Command : /sipp -sf terminator_4GEE_B2BUA_CR1_IAM_0001.xml -p 4449 -inf validate.csv -nd -trace_msg -trace_err -trace_logs SIPp is not injecting value ?5111313001? from validate.csv into terminator_4GEE_B2BUA_CR1_IAM_0001.xml. I have tried following parameters in scenario file: ?[field0]? è Value of [filed0] shows ?i? in the log file. ? [filed0] ? è Not working [field0 line=1] è Not working \?[field0]\? è SIPp is showing ?Segmentation Fault (core dumped)? error Note: Used SIPp 3.1 Version Please help me to resolve this issue. Thanks -Sumeet -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Problem about 2nd Via
You'll need two separate regexp match actions using the occurrence=1 and occurrence=2 tags. Charles Evgeny Miloslavsky emiloslav...@juniper.net wrote on 12/18/2008 01:26:52 AM: Hi Giuseppe! As a solution/workaround for this situation I would suggest you after INVITE request received by UAS, to assign first Via header to variable 1 and second Via header to variable 2. While sending responses from UAS put the first Via with value of variable 1 and second Via will have value of variable 2. As far as I understand it should look like: recv request=INVITE action ereg regexp=.* search_in=hdr header=Via: check_it=true assign_to=1,2 / /action /recv At sending response procedure simply put the following lines: Via: [$1] Via: [$2] I hope it will help. Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 / 7320 -Original Message- From: Giuseppe Roberti [mailto:j...@jnod.org] Sent: Wednesday, December 17, 2008 6:04 PM To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] Problem about 2nd Via Hi, i am testing a proxy using sipp but i have problem with Via headers. 10.0.0.1 is the proxy. 10.0.0.2 is the sipp server (-sn uas) 10.0.0.3 is the sipp client (-sn uac) I have noticed that the 2nd via added by sipp uas is threat incorrectly by the proxy but i don't know if it is my fault. Here the sip flow. 1) The sipp uac send to the proxy the INVITE: INVITE sip:serv...@10.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK From: sipp sip:s...@10.0.0.3:5060;tag=23603SIPpTag091 To: sut sip:serv...@10.0.0.1:5060 Call-ID: 1-23...@10.0.0.3 CSeq: 1 INVITE Contact: sip:s...@10.0.0.3:5060 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 186 v=0 o=user1 53655765 2353687637 IN IP4 10.0.0.3 s=- c=IN IP4 10.0.0.3 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 2) The proxy send the INVITE to sipp uas, adding the second Via header INVITE sip:serv...@10.0.0.2:5060 SIP/2.0 Record-Route: sip:10.0.0.1;lr=on;ftag=23603SIPpTag091;did=c04.37fe035 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK21dd.7d5758a6.0;rport Via: SIP/2.0/UDP 10.0.0.3:5060;rport=5060;received=10.0.0.3;branch=z9hG4bK From: sipp sip:s...@10.0.0.3:5060;tag=23603SIPpTag091 To: sut sip:serv...@10.0.0.1:5060 Call-ID: 1-23...@10.0.0.3 CSeq: 1 INVITE Contact: sip:s...@10.0.0.3:5060 Max-Forwards: 69 Subject: Performance Test Content-Type: application/sdp Content-Length: 187 v=0 o=user1 53655765 2353687637 IN IP4 10.0.0.3 s=- c=IN IP4 10.0.0.1 t=0 0 m=audio 50110 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 3) The sipp uas does not properly recognize (maybe) the 2nd Via and send back this (please note the line after the first Via) SIP/2.0 180 Ringing Record-Route: sip:10.0.0.1;lr=on;ftag=23603SIPpTag091;did=c04.37fe035 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK21dd.7d5758a6.0;rport, SIP/2.0/UDP 10.0.0.3:5060;rport=5060;received=10.0.0.3;branch=z9hG4bK From: sipp sip:s...@10.0.0.3:5060;tag=23603SIPpTag091 To: sut sip:serv...@10.0.0.1:5060;tag=15130SIPpTag011 Call-ID: 1-23...@10.0.0.3 CSeq: 1 INVITE Contact: sip:10.0.0.2:5060;transport=UDP Content-Length: 0 Is it my fault ? I'm using sipp from svn. -- Giuseppe Roberti j...@jnod.org -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/
Re: [Sipp-users] String manipulation with Sipp
The only way to do it is to use regular expressions. For anything else, you'll need to modify the source code. Adding an action is relatively straightforward. You need to change scenario.cpp to parse it, actions.hpp to define it, and call.cpp to actually do it. Charles Tuan Viet Nguyen tuanviet.ngu...@yahoo.fr 12/16/2008 11:28 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] String manipulation with Sipp Hello, Does anyone know if it is possible to manipulate characters with Sipp? For example to delete a character from a string variable, to replace a character , etc ... BR, Tuan Viet Nguyen -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] MAX rate of dialogs SIPp able to handle
It will depend on the hardware and complexity of your scenario, but 250 CPS should certainly be possible on a relatively modern Linux box [within the last 3-4 years] for the basic UAC/UAS scenario. The number of outstanding dialogs is mainly memory constrained, but there is some affect on CPU. How many are you looking at. Something in the range of 100,000 shouldn't be a problem (assuming they are just pausing). This assumes SIP only and no RTP. The best way to see that you're past the limit is to check the WatchDogMinor and WatchDogMajor counters, which fire if the system is unable to keep up. If it happens too many times, then SIPp will terminate. Charles Evgeny Miloslavsky [EMAIL PROTECTED] 12/09/2008 10:13 AM To [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject [Sipp-users] MAX rate of dialogs SIPp able to handle Hi All What is a maximal rate of dialogs SIPp is able to create? Is SIPp actually able to generate traffic of 250 INVITEs/second? Theoretically, I can set a really huge values while running it. But what are the actual numbers? What is a max number of dialogs SIPp is able to support? Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 / 7320 -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. The future of the web can't happen without you. Join us at MIX09 to help pave the way to the Next Web now. Learn more and register at http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] One Call-ID per scenario
If you only want a single scenario to execute; you can probably hack it up. Otherwise it will require a pretty serious design change. Charles Venkat Narasimhan [EMAIL PROTECTED] wrote on 12/01/2008 02:04:52 AM: Folks, I am looking for a hack/mod that lets SIPp(running an xml) accept/send SIP msgs irrespective of call-ID. Is this too much to expect/does it require just a small code change or a major design change in SIPp... Or has this been debated before ??? Any Responses will be deeply appreciated. Thanks in Advance Regards Venkat - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to process second invite with another call-id
You can create an out-of-call scenario that will create a new call for all unexpected messages using the -oocsf option. Charles Dhananjaya Reddy Eadala [EMAIL PROTECTED] 11/26/2008 11:04 PM To [EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net Subject Re: [Sipp-users] How to process second invite with another call-id I dont think this is possible with sipp. you might do this as follows: after 1st call is over, bring down sipp immediately and bring it up immediately with another scenario where you expect to recieve INVITE. but if you receive INVITE between stop and start of sipp, then it is gone. Dhana On Wed, Nov 12, 2008 at 12:24 AM, [EMAIL PROTECTED] wrote: Hi all, I'm sorry that I get in trouble. I wanna create a scenario like: sipp send a Invite with a call-id first, then sipp receive a 200 OK and sipp send a Ack. But sipp will receive a Invite with another call-id. I wanna sipp can process this received Invite ,send 180 and 200 OK. However , sipp only process the Invite as a Unknown message!!! Please tell me, how can I do this? Thank you. ZTE Information Security Notice: The information contained in this mail is solely property of the sender's organization. This mail communication is confidential. Recipients named above are obligated to maintain secrecy and are not permitted to disclose the contents of this communication to others. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. Any views expressed in this message are those of the individual sender. This message has been scanned for viruses and Spam by ZTE Anti-Spam system. - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] different pcap files for different calls?
You have to change scenario.cpp so that the play_pcap_audio argument instantiates a SendingMessage structure (use the action-setMessage function), and in call.cpp the createSendingMessage should be used where the CAction::E_AT_PLAY_PCAP_AUDIO is executed in executeAction(). It should involve less than a dozen lines of code. Charles Jan Rudinský [EMAIL PROTECTED] 11/21/2008 10:13 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] different pcap files for different calls? Hi, I'm using SIPp to generate SIP sessions with RTP media. How to select different pcap files for different calls? Using CSV database with paths to different pcaps doesn't work for play_pcap_audio attribute. However if I place log element just behind exec it prints the variable content right... nop action exec play_pcap_audio=[field1]/ log message=Path is [field1]/ /action /nop There is a related conversation, but without conclusion (http://osdir.com/ml/telephony.sipp.user/2007-06/msg00118.html). Does anyone know a hint? Thank you, Honza -- Ing. Jan Rudinsky RD Centre (RDC) for Mobile Applications Czech Technical University in Prague Cesnet z.s.p.o. [EMAIL PROTECTED] http://www.linkedin.com/in/rudinsky - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Variables as integer
You need to get a recent subversion trunk version, 3.1 does not have this support. Charles Shamik Mukhopadhyay -X (shmukhop - WIPRO at Cisco) [EMAIL PROTECTED] 11/13/2008 07:07 PM To mayamatakeshi [EMAIL PROTECTED], Charles P Wright/Watson/[EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net Subject RE: [Sipp-users] Variables as integer Hi Charles and Takeshi, I don't see ereg ... search_in=var variable=foo / happening in SIPp version 3.1. If you can provide me a pointer for the version where it works, it will be helpful. I need to increase Cseq:, which I'm not able to do in a loop. Thanks, Shamik From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mayamatakeshi Sent: Wednesday, September 17, 2008 10:38 AM To: Charles P Wright Cc: [EMAIL PROTECTED]; sipp-users Subject: Re: [Sipp-users] Variables as integer On Thu, Sep 18, 2008 at 2:31 AM, Charles P Wright [EMAIL PROTECTED] wrote: With recent trunk versions you can do: ereg ... search_in=var variable=foo / Oh! That's neat. Thanks a lot. mayamatakeshi [EMAIL PROTECTED] 09/17/2008 01:25 PM To Charles P Wright/Watson/[EMAIL PROTECTED] cc sipp-users sipp-users@lists.sourceforge.net, [EMAIL PROTECTED] Subject Re: [Sipp-users] Variables as integer On Thu, Sep 18, 2008 at 1:59 AM, Charles P Wright [EMAIL PROTECTED] wrote: The only way right now is to use a regular expression to parse it out as a string. Hi Charles, but is it possible to pass a variable to the ereg action? It seems search_in will only accept msg or hdr. I need a variable because I'll have to increment it. mayamatakeshi [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 09/17/2008 12:37 PM To sipp-users sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Variables as integer Hello, how can I insert a variable in a message but as an integer instead of a floating point number? For example, if I do: nop action assign assign_to=1 value=1 / /action /nop and try to use it like this ... CSeq: [$1] REGISTER ...the header will be sent as : CSeq: 1.00 REGISTER How can I make it to be sent as : CSeq: 1 REGISTER ? regards, takeshi - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPP Conditional Branching with If-Then-Else Variable Testing
David, If you get a recent trunk, you can use text names for labels and variables. I think you can even do it in 3.1. Charles David Adams [EMAIL PROTECTED] 11/04/2008 08:12 AM To Charles P Wright/Watson/[EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net Subject RE: [Sipp-users] SIPP Conditional Branching with If-Then-Else Variable Testing AWESOME! I had no idea that I could do the variable testing in a nop. This approach is working for me, however, I guess in the version that I have I must use integer labels instead of text ones, unless the example you provided was just to show me how to get an implementation of if-then-else with next test. Thanks very much Charles! Dave Adams. This email message and any attachments may be confidential and/or privileged to Nortel. If the reader of this message is not the intended recipient, you are hereby notified that any use, disclosure, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by replying to this message or by calling the sender and then destroying all copies of the message and any attachments. -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Monday, November 03, 2008 2:41 PM To: Adams, David (CAR:3P33) Cc: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] SIPP Conditional Branching with If-Then-Else Variable Testing if you set the variable you can do something like: nop next=ifclause test=variable / !-- else cluase stuff goes here -- nop next=endif / !-- if clause stuff goes here -- label id=endif / Charles David Adams [EMAIL PROTECTED] wrote on 11/03/2008 12:41:41 PM: Hi, I'm trying to build error-handling into a scenario with conditional- branching and variable testing. With the latest stable version of SIPP, I know I have the ability to use regexp to populate variables, and then in some actions such as send / receive I can test a variable and jump to a label after the send or receive action has been executed. What I'd like to be able to do is setup a variable, then based on whether the variable is set or not, then go to a different place in my scenario file. So far I have not been able to figure how to do this, if it's even possible. SIPP seems to support the Else part of the If-Then-Else with the variable testing and next arguments as part of send and receive operations. 1. Is there a way to do this in SIPP3.x? If not, 2. Could you direct me to the most logical place in the code that I would need to modify to build this feature? Thanks, Dave Adams. - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] sipp start_rtd and rtp issue!
Your nop is not terminated with a /nop (or alternatively nop rtd=true /). Charles ZHOU Gaofeng A [EMAIL PROTECTED] 10/16/2008 05:13 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] sipp start_rtd and rtp issue! Hi : who can help me?, Now I try to write a xml scenario simulated registrar server, but I cannot run it ok with the following error. How can I modify it? C:\Program Files\SIPpsipp -d 10 -i 135.251.25.238 -p 5060 -sf D:\call-flow\pack\ registrar.xml 2008-10-16 17:04:52:3341224147892.334594: You have started Response Time Duration 1, but have never stopped it!. C:\Program Files\SIPp registrar.xml: ?xml version=1.0 encoding=ISO-8859-1 ? !DOCTYPE scenario SYSTEM sipp.dtd !-- This program is free software; you can redistribute it and/or -- !-- modify it under the terms of the GNU General Public License as -- !-- published by the Free Software Foundation; either version 2 of the -- !-- License, or (at your option) any later version. -- !-- -- !-- This program is distributed in the hope that it will be useful, -- !-- but WITHOUT ANY WARRANTY; without even the implied warranty of -- !-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -- !-- GNU General Public License for more details. -- !-- -- !-- You should have received a copy of the GNU General Public License -- !-- along with this program; if not, write to the -- !-- Free Software Foundation, Inc., -- !-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -- !-- -- !-- Sipp default 'branchs' scenario. -- !-- -- scenario name=Registrar Server !--recv request=REGISTER start_rtd=true -- recv request=REGISTER start_rtd=true action ereg regexp=^5[0-9]*[0-9]$ search_in=hdr header=Contact: check_it=true assign_to=4/ ereg regexp=^\?.*\?$ search_in=hdr header=From: check_it=true assign_to=5/ /action /recv send ![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: sip:[EMAIL PROTECTED]:[local_port];transport=[transport];expires=3600 Content-Length: 0 Allow-Events: reg P-Associated-URI: [$5] Path: sip:pcsf-stdn.imsgroup0-000.fs5k8.shanghai.com:5060;lr;bidx=0 ]] /send nop rtd=true !-- definition of the response time repartition table (unit is ms) -- ResponseTimeRepartition value=1000, 1040, 1080, 1120, 1160, 1200/ !-- definition of the call length repartition table (unit is ms) -- CallLengthRepartition value=1000, 1100, 1200, 1300, 1400/ /scenario Thanks! Jack - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] error: ?INT_MAX? was not decl ared in this scope
On RedHat it is in /usr/include/limits.h, which is part of the glibc-headers package. Charles Antoine [EMAIL PROTECTED] 10/16/2008 02:51 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] error: ?INT_MAX? was not declared in this scope I am trying to install SIPp on a Linux Mandriva 2009 (did it previously on Mandriva 2008 without any problems) however I am getting the following error after running make pcapplay: scenario.cpp:900: error: ?INT_MAX? was not declared in this scope Same error if I just do a make. Anyone has seen this error before? [EMAIL PROTECTED] sipp.svn]$ make pcapplay make OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_PCAPPLAY=send_packets.o prepare_pcap.o PCAPPLAY_LIBS=-lpcap PCAPPLAY=-DPCAPPLAY sipp make[1]: Entering directory `/home/antoine/SIPp/sipp.svn' g++ -D__LINUX -pthread -DSVN_VERSION=\unknown\ -DPCAPPLAY -I. -I/usr/include/openssl -c -o scenario.o scenario.cpp scenario.cpp: In constructor ?scenario::scenario(char*, int)?: scenario.cpp:900: error: ?INT_MAX? was not declared in this scope scenario.cpp: In function ?CSample* parse_distribution(bool)?: scenario.cpp:1054: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1056: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1058: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1060: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1062: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1064: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1066: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1068: warning: deprecated conversion from string constant to ?char*? scenario.cpp: In member function ?void scenario::getActionForThisMessage()?: scenario.cpp:1320: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1320: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1321: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1321: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1411: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1443: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1446: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1449: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1452: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1455: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1458: warning: deprecated conversion from string constant to ?char*? scenario.cpp: At global scope: scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:1792: warning: deprecated conversion from string constant to ?char*? scenario.cpp:3161: warning: deprecated conversion from string constant to ?char*? make[1]: *** [scenario.o] Error 1 make[1]: Leaving directory `/home/antoine/SIPp/sipp.svn' make: *** [pcapplay] Error 2 - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net
Re: [Sipp-users] Poisson call arrival rate code in IMS Bench SIPp
I've done something similar to this in the regular SIPp using third party call control. Essentially, I had a controller that would pause exponentially in a loop, and then send a 3PCC kick-off message to a UAC. The easier way is probably to modify the opentask.cpp class to support a Poisson arrival given the regular rate parameter and a new command line flag. Charles Muhammad Ali [EMAIL PROTECTED] 10/14/2008 10:29 AM Please respond to [EMAIL PROTECTED] To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Poisson call arrival rate code in IMS Bench SIPp Hi, Which part of the code in IMS Bench SIPp is used to generate Poisson call arrival rate? Can it be appended with normal SIPp code? Any help in this regard will be helpful. I need to implement Poisson call arrival rate in a VoIP experiment. Best regards M Ali - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Why doesn´t SIPp re-send the ACK wh en it receive a lot of messages 200 OK?
You should include the output of -trace_calldebug in your message. Charles ZiLi0n [EMAIL PROTECTED] wrote on 10/07/2008 04:33:10 PM: SIPp should retransmit the ACK if the retransmitted 200 is identical. You should enable -trace_msg and -trace_calldebug and see what the logs tell you. The retrasnmitted meesages 200 OK are identicals. I´m using SIPp with Asterisk.Asterisk show this errors: Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 for seqno 1620 (Critical Response) Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Hanging up call 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 no reply to our critical packet. A solution? Thanks Charles ZiLi0n [EMAIL PROTECTED] 10/05/2008 02:26 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a lot of messages 200 OK? I am testing my Asterisk Server. This is my configuration: SIPpCLIENT --- Asterisk Server --- SIPpSERVER If the simultaneous calls is less than 300 the test is OK, but when the simultaneous calls is approximately 350 calls, the calls hang up: Asterisk sends to SIPpCLIENT the message 200 OK. SIPpCLIENT sends to Asterisk the message ACK. Asterisk re-sends to SIPpCLIENT the message 200 OK. SIPpCLIENT doesn´t re-send the message ACK to Asterisk. I think that message ACK is lost... but the cpu load Asterisk Server is approximately 50% and the net status is perfect. Why does not SIPpCLIENT re-send the ACK to Asterisk? Asterisk needs the ACK for complete the call Thank´s La cartera, las gafas. ¿te falta algo? Ahora llévate Messenger en tu móvil - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer'schallenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users Ahora llévate lo mejor de MSN y Windows Live, en tu móvil - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] From-Tag as alpha-numeric string
The easiest way would be to use an injection file. Charles Evgeny Miloslavsky [EMAIL PROTECTED] wrote on 10/05/2008 09:34:33 AM: Hi All Is there any possibility to set From-tag of INVITE request sent from SIPp as random length alpha-numeric string with possibility to include both BNF allowed and not allowed chars and not as [pid] SIPpTag00[call_number] as it appears at uac.xml default scenario. For example how do I set From-Tag something like tag=EBJ[9p^yeB Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 / 7320 - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Why doesn´t SIPp re-send the ACK wh en it receive a lot of messages 200 OK?
SIPp should retransmit the ACK if the retransmitted 200 is identical. You should enable -trace_msg and -trace_calldebug and see what the logs tell you. Charles ZiLi0n [EMAIL PROTECTED] 10/05/2008 02:26 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a lot of messages 200 OK? I am testing my Asterisk Server. This is my configuration: SIPpCLIENT --- Asterisk Server --- SIPpSERVER If the simultaneous calls is less than 300 the test is OK, but when the simultaneous calls is approximately 350 calls, the calls hang up: Asterisk sends to SIPpCLIENT the message 200 OK. SIPpCLIENT sends to Asterisk the message ACK. Asterisk re-sends to SIPpCLIENT the message 200 OK. SIPpCLIENT doesn´t re-send the message ACK to Asterisk. I think that message ACK is lost... but the cpu load Asterisk Server is approximately 50% and the net status is perfect. Why does not SIPpCLIENT re-send the ACK to Asterisk? Asterisk needs the ACK for complete the call Thank´s La cartera, las gafas. ¿te falta algo? Ahora llévate Messenger en tu móvil - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] implementing SIP proxy and redirect Servers
You can also use OpenSER for a SIP proxy/registrar. If you are particularly motivated, it think it would be possible to get SIPp to behave like these servers; but it is probably not worth the effort. Charles Jeff Wright [EMAIL PROTECTED] 10/03/2008 09:26 AM To amar mahmoud [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] implementing SIP proxy and redirect Servers We use sipX as a SIP proxy and registrar in our test lab. It is free, easy to install and configure. http://www.sipfoundry.org/sipX/ Jeffrey Wright System Test Engineering Manager Aztek Networks, Inc. -Original Message- From: amar mahmoud [mailto:[EMAIL PROTECTED] Sent: Fri 10/3/2008 6:40 AM To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] implementing SIP proxy and redirect Servers Hi, I want to build test bed contains SIP user agents and also proxy and redirect servers, I can use SIPp for user agents, but what about servers. anyone who has idea about which easiest tool should I use with combination with SIPp causing no problem, I need only the main functionalities of those servers. Amar _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live. http://clk.atdmt.com/MRT/go/msnnkwxp1020093185mrt/direct/01/ - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Call Rate Decreases with time
You can try -recv_timeout Ns. Charles Ricardo Fernandes [EMAIL PROTECTED] 09/26/2008 04:56 AM To sipp-users@lists.sourceforge.net, Charles P Wright/Watson/[EMAIL PROTECTED] cc Subject Call Rate Decreases with time Hello, I am facing a problem regarding the call rate option. I define a call rate of 12 call a second to my server. After a while the call rate decreases in sipp. I will show the counter OutgoingCall(C) to explain: 1 hour - 4512 2 hour - 7287 3 hour - 9747 4 hour - 11414 5 hour - 12922 6 hour - 14266 7 hour - 16220 8 hour - 18013 9 hour - 18562 10 hour - 19273 11 hour - 20029 12 hour - 20563 13 hour - 21000 I think the problem cames from the calls that are not completed and abandoned by my server. In 21000 calls i have 30 calls that considered abandoned by sipp(Aborted call with Call-ID '[EMAIL PROTECTED]'.) The problem is, i think, is that sipp consideres that these calls are active until the end, and because of that does not make new calls because 30 calls are always active until the end. I have tried to use the -deadcall_wait to see if sipp release these calls after a while, but it did not make a difference. Now the million dollar question: Is there a way for sipp to release these calls after a certain timeout if no response comes from the server? I am using the lastest version(sipp-win32-2008-08-26.exe) of the unstable downloads on a windows XP Professional Service Pack 2. TIA Ricardo Fernandes - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Different port for remote side
SIPp should be able to receive and process the message from a different port. I would use a combination of packet capture and -trace_msg to make sure the packet arrives at SIPp. Charles Romain Gautier [EMAIL PROTECTED] wrote on 09/24/2008 04:37:54 AM: Hi, I am trying to use SIPp as an UAC, using a scenario. My problem is that SIPp does not seem to recognize the SIP responses. Indeed the 180 trying is not taken into account by SIPp. The remote part use a different port in order to send its response, is it a problem? Is there any turn-around? SIPp(5060) Remote INVITE (port 5060) port 5060 port 5060 TRYING from port (dynamic port) Thank you for your answer. Cordially - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Issue faced with updating filed value of csv injection file
No. SIPp's internal timing loop will go haywire trying to catch up when you resume it. Charles Madiha Shahid [EMAIL PROTECTED] 09/24/2008 01:05 AM To Peter Higginson [EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net Subject Re: [Sipp-users] Issue faced with updating filed value of csv injection file Thanks Peter for the suggestion. Thats right, the scenario I'm using works for one call only. Would it be a good idea to pause the SIPP process using 'kill -SIGSTOP' command on linux and resume it after the media transfer gets completed by using the 'kill -SIGCONT' command? Regards, Madiha On Wed, Sep 24, 2008 at 1:36 AM, Peter Higginson [EMAIL PROTECTED] wrote: Madiha, The mechanism you have described looks like it only works with one call. If that is the case you could exit SIPP (saving any context and the Call-ID of course) and re-enter it to continue the call after the media is done. The alternative we did at Newport Networks was to start and stop the external media generator from the SIPP process. That method will (and did) work for multiple simultaneous calls and you then use something like a pause to control the length of the media generation. Peter Higginson Date: Tue, 23 Sep 2008 21:35:33 +0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] CC: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] Issue faced with updating filed value of csv injection file Hi, Thanks for the reply Charles. Does anyone know a workaround to this problem. Is there a way to induce a variable pause at the server side SIPp such that the the file execution of the server side resumes only after the media transfer gets completed. Regards, Madiha On Tue, Sep 23, 2008 at 4:49 PM, Charles P Wright [EMAIL PROTECTED] wrote: You can not update the value of CSV fields after starting SIPp. Charles Madiha Shahid [EMAIL PROTECTED] 09/23/2008 03:02 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Issue faced with updating filed value of csv injection file Hi all, Description: I am writing a scenario in SIPp that allows media transfer between calls using an external utility (Gstreamer). The external utility gets called by running it through exec command.I want to induce a pause at the sender side so that media transfer gets completed before further messages can be tranfered between SIPp client and server. This is how Im trying to do it. I use the -inf switch and provide a csv file as input to the server side sipp command The [field0] in this csv file has vale 1. When file transfer gets completed, value '1' written in this file is replaced with value '10' as written by an external application. The SIPp server, keeps monitoring the [field0] value to check if the the file has been updated so that it can proceed further. However, even though the value in the csv file is replaced, it is not updated in the [field0]. [field0] still has the old value which keeps the scenario in a loop for ever. Please let me know if this is expected? Is there a workaround to this problem? Thanks, Madiha Here is the part of the code at the server side that produces this issue: ** ** nop action exec command=./gst-sender.sh/ /action /nop label id=8/ nop action log message=entered label 8/ /action /nop pause milliseconds=1/ nop action !-- Assign the value in field0 of the CSV file to a $3. -- assignstr assign_to=3 value=[field0] / log message=Value written in file is [$3]/ todouble assign_to=4 variable=3 / log message=Value written in file converted is [$4]/ test assign_to=5 variable=4 compare=not_equal value=10 / log message=Result of compare is [$5]/ /action /nop nop next=8 test=5/ nop action log message=exiting label 8/ - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users Get Hotmail on your mobile from Vodafone Try it Now! - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing
Re: [Sipp-users] Different port for remote side
You should post your message trace and error trace to the list. Charles Romain Gautier [EMAIL PROTECTED] wrote on 09/24/2008 10:13:16 AM: Thank you for your reply. Indeed SIPp logs successfullly the TRYING within the embedded uac scenario. Nevertheless, using my own scenario, SIPp logs the TRYING in the errors log file: the TRYING cannot be mapped to a known SIPp call, although the Call-ID is correct. Should it be an encoding issue of my scenario file? Cdt Romain 2008/9/24 Charles P Wright [EMAIL PROTECTED] SIPp should be able to receive and process the message from a different port. I would use a combination of packet capture and -trace_msg to make sure the packet arrives at SIPp. Charles Romain Gautier [EMAIL PROTECTED] wrote on 09/24/2008 04:37:54 AM: Hi, I am trying to use SIPp as an UAC, using a scenario. My problem is that SIPp does not seem to recognize the SIP responses. Indeed the 180 trying is not taken into account by SIPp. The remote part use a different port in order to send its response, is it a problem? Is there any turn-around? SIPp(5060) Remote INVITE (port 5060) port 5060 port 5060 TRYING from port (dynamic port) Thank you for your answer. Cordially - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Issue faced with updating filed value of csv injection file
You can not update the value of CSV fields after starting SIPp. Charles Madiha Shahid [EMAIL PROTECTED] 09/23/2008 03:02 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Issue faced with updating filed value of csv injection file Hi all, Description: I am writing a scenario in SIPp that allows media transfer between calls using an external utility (Gstreamer). The external utility gets called by running it through exec command.I want to induce a pause at the sender side so that media transfer gets completed before further messages can be tranfered between SIPp client and server. This is how Im trying to do it. I use the -inf switch and provide a csv file as input to the server side sipp command The [field0] in this csv file has vale 1. When file transfer gets completed, value '1' written in this file is replaced with value '10' as written by an external application. The SIPp server, keeps monitoring the [field0] value to check if the the file has been updated so that it can proceed further. However, even though the value in the csv file is replaced, it is not updated in the [field0]. [field0] still has the old value which keeps the scenario in a loop for ever. Please let me know if this is expected? Is there a workaround to this problem? Thanks, Madiha Here is the part of the code at the server side that produces this issue: ** ** nop action exec command=./gst-sender.sh/ /action /nop label id=8/ nop action log message=entered label 8/ /action /nop pause milliseconds=1/ nop action !-- Assign the value in field0 of the CSV file to a $3. -- assignstr assign_to=3 value=[field0] / log message=Value written in file is [$3]/ todouble assign_to=4 variable=3 / log message=Value written in file converted is [$4]/ test assign_to=5 variable=4 compare=not_equal value=10 / log message=Result of compare is [$5]/ /action /nop nop next=8 test=5/ nop action log message=exiting label 8/ - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Caller scenario sends out REGISTER packets without respecting -users or -l flag
There is no way to limit transactions or requests; only calls (either with -l or -users). If your call has only one concurrent transaction (probably the only way for SIPp to work correctly); then the number of calls is an upper bound on transactions. You can disable retransmissions with -nr to prevent more than one request in the same transaction; but that is not going to give you an accurate workload. If a call fails (i.e. the INVITE is never replied to); then that call is replaced with a new one that sends register. You can limit the total number of calls with -m 100. Charles Manish Sapariya [EMAIL PROTECTED] wrote on 09/22/2008 06:16:19 AM: Hi All, I am trying to create a work load where in I want to have 100 max established calls after the system has reached count of 100 calls. My caller scenario is approximately as follows: === Send Register Expect proxy auth Send Register with auth Expect 200 OK Send Invite Expect Proxy auth Send Invite with auth expect OK play pcap file wait for the duration of pcap file Send Bye Expect OK = If my server under test sends the response to both register and Invite within time for all 100 requests, everything works just fine. However, if for some reason, my server fails to send reply to some of the invite packets, then sipp keeps on sending register packets irrespective of how many total register packet it has sent. In this way it keeps bombarding my server with register packets, and server fails to send the reply to the invite packet. I am sure there is a problem with server, however question to the list is that, Is it possible to tell sipp that keep at the max 100 outstanding register request or invite request. I tried using -l and -users option. However both of this do not take un-acknowledged register and invite request into account. Please let me know if I need to provide more info or clarification. I can share the scenario and the exact command line it that helps. Thanks and Regards, Manish - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIP-I message format in sipp
This is applied as revision 536. It would be great if we had a nice string structure throughout the code so that we could handle non-null terminated strings both on send and receive for all types of messages; but this is a good start. Charles Andy Aicken [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 09/09/2008 07:04 PM To 'darshan b n' [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] SIP-I message format in sipp Hi Darshan, I created a patch for handling SIP-I messages, as the current message handling in SIPp treats everything as a string so doesn?t handle ISUP message bodies that contain a binary \x00. This ends up being treated as a string termination resulting in message gets truncated. The patch is available at: https://sourceforge.net/tracker/?func=detailatid=637566aid=1965508group_id=104305 It needs more rigorous testing but worked ok for me with the type of functionality I was using. Regards Andy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of darshan b n Sent: 04 September 2008 12:30 To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] SIP-I message format in sipp Hi all , i want know how to create a SIP-I message in sipp please respond with a sample message format Thanks darshan On 04/09/2008, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Send Sipp-users mailing list submissions to sipp-users@lists.sourceforge.net To subscribe or unsubscribe via the World Wide Web, visit https://lists.sourceforge.net/lists/listinfo/sipp-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Sipp-users digest... Today's Topics: 1. Re: Force source IP source Port at IP layer (Klaus Darilion) 2. sipp remote (RTP) port handling (Jan Rudinsk?) -- Message: 1 Date: Thu, 04 Sep 2008 10:05:02 +0200 From: Klaus Darilion [EMAIL PROTECTED] Subject: Re: [Sipp-users] Force source IP source Port at IP layer To: Cyrille OLIVIER [EMAIL PROTECTED] Cc: sipp-users@lists.sourceforge.net Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed FYI: If you want to change the src IP you can also use this patch: https://sourceforge.net/tracker/?func=detailatid=637566aid=1823593group_id=104305 klaus Cyrille OLIVIER schrieb: Dear sipp-users, Again, I asked my requests about SIPp client using TCP: Is it possible to force sipp to use specific IP source Port source, at IP layer, for send messages when TCP with single socket (option '-t t1' used) ? I tried many things: -bind_local: seems unuseful. -i x.x.x.x -p options: it's only for some SIP headers but not for IP packet header. send -source_ip=x.x.x.x -source_port= for INVITE message look for this subject in mailing list archives ... Currently, I don't know which other workaround or things to do :( I would really appreciate any help about that Thanks a lot, BR, Cyrille Discutez gratuitement avec vos amis en vid?o ! T?l?chargez Messenger, c'est gratuit ! http://www.windowslive.fr/majmessenger.asp - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Message: 2 Date: Thu, 04 Sep 2008 12:32:12 +0200 From: Jan Rudinsk? [EMAIL PROTECTED] Subject: [Sipp-users] sipp remote (RTP) port handling To: sipp-users@lists.sourceforge.net Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-2 Hi, I'm using SIPp to generate a call with RTP media. Media are sent to remote side, recorded and sent back. However SIPp sends media to a different remote port than offered by the remote side. SIPp: SIP INVITE with SDP m=audio 6000 RTP/AVP 0 Remote: 200 OK with SDP m=audio 18436 RTP/AVP 0 101 SIP:RTP incoming on 6000(OK) Remote: RTP incoming on 1843(instead of 18436) Attached: scenario graph, packet capture Does anyone know the solution? Thank you, JaR -- Ing. Jan Rudinsky Czech Technical University in Prague Cesnet z.s.p.o. RD Centre (RDC) for Mobile Applications [EMAIL PROTECTED]
Re: [Sipp-users] Variables as integer
The only way right now is to use a regular expression to parse it out as a string. Charles mayamatakeshi [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 09/17/2008 12:37 PM To sipp-users sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Variables as integer Hello, how can I insert a variable in a message but as an integer instead of a floating point number? For example, if I do: nop action assign assign_to=1 value=1 / /action /nop and try to use it like this ... CSeq: [$1] REGISTER ...the header will be sent as : CSeq: 1.00 REGISTER How can I make it to be sent as : CSeq: 1 REGISTER ? regards, takeshi - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Variables as integer
With recent trunk versions you can do: ereg ... search_in=var variable=foo / Charles mayamatakeshi [EMAIL PROTECTED] 09/17/2008 01:25 PM To Charles P Wright/Watson/[EMAIL PROTECTED] cc sipp-users sipp-users@lists.sourceforge.net, [EMAIL PROTECTED] Subject Re: [Sipp-users] Variables as integer On Thu, Sep 18, 2008 at 1:59 AM, Charles P Wright [EMAIL PROTECTED] wrote: The only way right now is to use a regular expression to parse it out as a string. Hi Charles, but is it possible to pass a variable to the ereg action? It seems search_in will only accept msg or hdr. I need a variable because I'll have to increment it. mayamatakeshi [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 09/17/2008 12:37 PM To sipp-users sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Variables as integer Hello, how can I insert a variable in a message but as an integer instead of a floating point number? For example, if I do: nop action assign assign_to=1 value=1 / /action /nop and try to use it like this ... CSeq: [$1] REGISTER ...the header will be sent as : CSeq: 1.00 REGISTER How can I make it to be sent as : CSeq: 1 REGISTER ? regards, takeshi - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] UDP destination port in server mode
Marc, If you can extract the port from the header, you can use the new, but as of yet undocumented setdest action: Something like: nop action assignstr assign_to=url value=[next_url] / ereg regexp=sip:(.*)@([0-9A-Za-z\.]+):([0-9]+);transport=([A-Z]+) search_in=var check_it=true assign_to=dummy,name,host,port,transport variable=url / warning message=HOST: [$host], PORT: [$port], TRANSPORT: [$transport] / setdest host=[$host] port=[$port] protocol=[$transport] / log message=[$host];[$port];[$transport];sip:[EMAIL PROTECTED]:[$port] / /action /nop Charles [EMAIL PROTECTED] wrote on 09/05/2008 10:31:22 AM: Hello all, I'm struggling to get sipp send out the response packets to the right UDP port. I'm using a pretty recent SIPp dated 20080723. SIPp is listening for register requests on port 5060 (-t u1 -p 5060) and answers them with a 200 OK. The register request messages arrive from a fixed source address (S- IP), with a variable UDP source port and have internally a Via:- header specifying S-IP:5060, so the responses are expected there, and not an the variable source port number. Not specifying anything special on the command line, responses go from SIPp:5060 to S-IP:source-port, instead of the address mentioned in the Via:. I've seen somewhere a mention that it can follow the via, but didn't find anything on that in documentation of source, so I think it is not in. Anybody know more of this? As in my case the destination is the fixed S-IP:5060, I tried specifying this with the '-rsa' remote sending address option. Using this option has a clear effect on the behaviour: SIPp now sends the message from SIPp:variable-high-port to S-IP:source-port instead of using 5060 as source port. This seems very strange to me, the SIPp source port gets variable, but the specified sending address:port is not used, also not when giving another IP address as rsa- destination. So, using the rsa-option has an effect, but not really the expected one. Anybody knowing how to solve this problem with SIPp? Best regards, MarcVD (-: from Marc VAN DIEST (BELGACOM) ;-) - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] inserting timestamp of messages
You can use -trace_logs and the [timestamp] keyword inside of a log message=[timestamp] / action. Charles [EMAIL PROTECTED] wrote on 09/09/2008 09:01:44 PM: Hi, when using trace_msg options I can get the message with its timestamp, But I need to extract only the timestamp in a CVS file i have tried to execute external command through action but got problem with that. Anyone who can help with that. thanks, Amar Ahmed Want to do more with Windows Live? Learn ?10 hidden secrets? from Jamie. Learn Now - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] can SIPp execute shell command like echo in background mode?
I suspect your use of double quotes within the quoted string is causing the problem. Charles [EMAIL PROTECTED] wrote on 09/12/2008 05:51:24 AM: Hi all, I am not able to log the status of the test in the log file when sipp is running in the background mode. It is able to log the status when it runs in the normal mode. I am using following action in the xml file: nop action exec command=echo Test case ID 101 is pass pass.log/ exec int_cmd=stop_now/ /action /nop This code doesn't log the status in the pass.log file when sipp is running in the background mode. Please help me to solve this problem. Thanks and Regards -Sumeet - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Arithmetic expression
You need to convert the string from the header into a double first. Try this before your increment: todouble assign_to=6 variable=6 / Charles [EMAIL PROTECTED] wrote on 09/02/2008 10:56:18 AM: HI There, I am trying to use the following statement in my sipp script and its not working, as I don't see the variable value incremented by 2, can you tell me what I am doing wrong? (attached is the script). I am using SIPP R3.0 recv request='ACK' rtd='true' crlf='true' action ereg regexp='.*' search_in='hdr' header='To:' check_it='true' assign_to='3'/ ereg regexp='.*' search_in='hdr' header='From:' check_it='true' assign_to='4'/ ereg regexp='([[:alnum:]]*) ([[:alnum:]]*)' search_in='hdr' header='CSeq:' check_it='true' assign_to='5,6'/ log message='6 is [$6]'/ add assign_to='6' value='2'/ //THIS ONE IS NOT WORKING log message=' value of 6 is now [$6]'/ /action /recv From, Nazia Hussain See what people are saying about Windows Live. Check out featured posts. Check It Out![attachment sip_uas_test.xml deleted by Charles P Wright/Watson/IBM] - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Error in Pause statement.
pause is not an action, it is a top-level element like recv Charles [EMAIL PROTECTED] wrote on 09/08/2008 10:13:08 PM: All, I'm using SIPp3.1. My SIPp is erroring at the following lines - can someone provide any guidance as to what the proper syntax should be? This xml file works okay with sipp2.0 recv response=500 optional=true action pause milliseconds=7 next=3/ /action /recv Error: 2008-09-08 21:12:00:0091220926320.009167: Unknown action: pause. thank you, Kalpesh. Katwala - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] RTP audio/video ports
Scott, I don't see a reason to have the fixed 2 port offset. I would be amenable to a patch if no one else objects. Charles Scott Oaks [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 09/02/2008 03:34 PM Please respond to [EMAIL PROTECTED] To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] RTP audio/video ports We are having problems again with the way sipp chooses rtp ports. sipp will search for a free port for RTP audio from 6000 to 6099 (by default). Having found a free port for the audio, it will then always use that port + 2 for the RTP video socket. But there's no assurance that port will be free, and the port checking loop logic doesn't extend to that: if the video port fails, sipp fails. Presumably we should fix this by binding both ports within the loop that does the port searching -- is there any problem with that? For people who don't want to use RTP, is there a reason why this code couldn't be skipped altogether (the code sort-of implies there used to be such an option, but now all those sockets are in an if(1) block). -Scott - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] multi line header
Roman, You can try the following: Subject: this is a \x20multi-line: Which will include the space before multi-line:. Charles [EMAIL PROTECTED] wrote on 09/04/2008 10:37:06 AM: Hi , I am trying to make sipp to send message with the multi line header like that: Subject: this is a \r\n multi-line: but the sipp is ignoring all the spaces in the second line, according to the spec the second line should begin with at least one space, there is no option to insert any special characters either. If anyone have an idea how I can insert space in the second line please advice. Thanks in advance ___ Roman Mandeleil Software engineer , SIP Container IBM Software Group, Israel Software Lab Office: +972-8-9401228 ext. 113 Mobile: +972-54-7644377 e-mail: [EMAIL PROTECTED] http://w3n.haifa.ibm.com/ilsl/rtc/infrastructure.html - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] RegExp: Finding nth occurence?
No. Charles [EMAIL PROTECTED] wrote on 08/29/2008 08:12:03 AM: Is there a way to find the nth occurance of a regexp in msg for example, I may need to find how many times the following match occurs ereg regexp=a=cparmin: search_in=msg assign_to=15/ Send instant messages to your online friends http://uk.messenger.yahoo.com - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] string manipulation for variables in sipp
There is no string manipulation, you'll need to change your regular expression to exclude the . Charles [EMAIL PROTECTED] wrote on 08/22/2008 06:10:29 AM: Hi I'm experiencing the following problem. When I receive the INVITE message, I'm able to store the Contact header in a variable, for example the variable 1. This variable contains also the character and at the beginning and at the end of the string. Since I need to reuse the value of the Contact header as Request-URI of the BYE message, I need to drop the character and from the variable 1. How can I do? Where I can find some commands about string manipulation for variables in sipp? thanks to all best regards corrado orlando - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Unexpected-message.
You should look at your error logs. Your system under test is likely generating error messages, indicating it is beyond its capacity. Charles Thekkedath, Sooraj (Sooraj) [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 08/22/2008 01:24 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Unexpected-message. Hi I am using SIPP for performance testing. In high call rate I am seeing more unexpected messages are coming. My call rate is 500 calls per second and call hold time is 200 sec. I think in this call load sipp is generating lot of unexpected messages, how can I avoid this? Thanks Sooraj Thekkedath Alcatel-Lucent Software Engineer Bangalore , India Phone : +91-80-3983-2180 Mobile : +91-9880537131 email: [EMAIL PROTECTED] - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] T.38 media type m=image
There is no support in SIPp right now. To add it you'll need to change call.cpp to support m=image instead of m=audio and m=video. Search for PAT_AUDIO and PAT_VIDEO in call.cpp. Charles Patrick Miccio [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 08/19/2008 08:29 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] T.38 media type m=image Hello everyone, I used google and searched the mailing list, but couldn't find any answers :( I am trying to recreate a fax call with Sipp, unfortunately I get the following error: media_port keyword with no audio or video on the current line (m=image ). Is there any workaround? here is the SDP information that causes the problem: ... ... Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIP Call c=IN IP[media_ip_type] [media_ip] t=0 0 m=image [media_port] udptl t38 a=T38FaxFillBitRemoval:0 a=T38FaxMaxBuffer:200 a=T38FaxTranscodingJBIG:0 a=T38FaxTranscodingMMR:0 a=T38FaxUdpEC:t38UDPRedundancy a=T38MaxBitRate:14400 a=T38FaxVersion:0 a=T38FaxMaxDatagram:72 a=T38FaxRateManagement:transferredTC - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Some errors on the log
The watchdog error means that SIPp set a timer for 400ms in the future, but it took 538 ms for it to come around. If this happens often, it is an indication that your SIPp machine is overloaded. Charles [EMAIL PROTECTED] wrote on 08/19/2008 11:14:10 AM: hi, I am getting some errors on my _errors.log file that shows as follows 1219140649.897846: The minor watchdog timer 500ms has been tripped (538), 109 trips remaining.. 1219140649.915473: send_packets.c: sendto failed with error: Invalidargument.. Can anyone help me to resolve the same? My calls are failing on my script. thanks, naresh Did you know? You can CHAT without downloading messenger. Click here - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] T.38 media type m=image
This should be fixed in trunk. In earlier versions change char number[X] in get_remote_port_media to char number[7], where X 7. Charles [EMAIL PROTECTED] wrote on 08/19/2008 12:06:21 PM: hey everyone, looks like I found a related problem: I send this SDP info in my INVITE: m=image 6000 udptl t38 I receive this SDP info in the 200 OK: m=image 50682 udptl t38 after that I execute: nop action exec play_pcap_video= t38.pcap/ /action /nop according to wireshark, sipp will send the RTP to port 5068, so it misses 1 digit in the media port, how can that happen? User Datagram Protocol, Src Port: 6000, Dst Port: 5068 cheers, Patrick. If your job permits; it would be great if you could post your patch to the list for others to use. yeah :) attached is the T.38 patch for sipp.3.1 you need to exec the pcap file with the play_pcap_video command! cheers, Patrick Charles [EMAIL PROTECTED] wrote on 08/19/2008 10:07:16 AM: hey, There is no support in SIPp right now. To add it you'll need to change call.cpp to support m=image instead of m=audio and m=video. Search for PAT_AUDIO and PAT_VIDEO in call.cpp. that worked like a charm :) THX, Patrick. Charles Patrick Miccio [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 08/19/2008 08:29 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] T.38 media type m=image Hello everyone, I used google and searched the mailing list, but couldn't find any answers :( I am trying to recreate a fax call with Sipp, unfortunately I get the following error: media_port keyword with no audio or video on the current line (m=image ). Is there any workaround? here is the SDP information that causes the problem: ... ... Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIP Call c=IN IP[media_ip_type] [media_ip] t=0 0 m=image [media_port] udptl t38 a=T38FaxFillBitRemoval:0 a=T38FaxMaxBuffer:200 a=T38FaxTranscodingJBIG:0 a=T38FaxTranscodingMMR:0 a=T38FaxUdpEC:t38UDPRedundancy a=T38MaxBitRate:14400 a=T38FaxVersion:0 a=T38FaxMaxDatagram:72 a=T38FaxRateManagement:transferredTC - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Some errors on the log
The timer is set to go off every 400ms. If it takes more than 3000ms for it to get triggered that is a major trip; 10 of those are a fatal error. If it takes more than 500ms, it is a minor trip; 120 count as a fatal error. If no firings occur for 10 minutes the counts are reset. These thresholds can be adjusted with command line options, and defaults are in sipp.hpp. Charles extern unsigned long watchdog_interval_DEFVAL(400); extern unsigned long watchdog_minor_threshold _DEFVAL(500); extern unsigned long watchdog_minor_maxtriggers _DEFVAL(120); extern unsigned long watchdog_major_threshold _DEFVAL(3000); extern unsigned long watchdog_major_maxtriggers _DEFVAL(10); extern unsigned long watchdog_reset _DEFVAL(60); Nicholas SHI [EMAIL PROTECTED] wrote on 08/19/2008 11:12:52 PM: Hi Charles, Sorry to hijack your response. Just want to make sure how you get the watchdog timer is 400ms other than 500ms. I wonder if this is typo error or something magic defined in other place. Thank you! Per log here: 1219140649.897846: The minor watchdog timer 500ms has been tripped (538), 109 trips remaining.. I thought the threashold is 500ms at first. Regards, Nicholas SHI --- Qingdao, China 2008/8/19 Charles P Wright [EMAIL PROTECTED]: The watchdog error means that SIPp set a timer for 400ms in the future, but it took 538 ms for it to come around. If this happens often, it is an indication that your SIPp machine is overloaded. Charles [EMAIL PROTECTED] wrote on 08/19/2008 11:14:10 AM: hi, I am getting some errors on my _errors.log file that shows as follows 1219140649.897846: The minor watchdog timer 500ms has been tripped (538), 109 trips remaining.. 1219140649.915473: send_packets.c: sendto failed with error: Invalidargument.. Can anyone help me to resolve the same? My calls are failing on my script. thanks, naresh Did you know? You can CHAT without downloading messenger. Click here - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer'schallenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Nicholas SHI - Tel: +86 131 5638 9168 Email: [EMAIL PROTECTED] Location: Qingdao, China URL: http://picasaweb.google.com/shixiaomu - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] check_it
Nop is descibed in the documentation: http://sipp.sourceforge.net/doc3.0/reference.html#Create+your+own+XML+scenarios The strcmp and condexec attributes are new and are not yet documented (I only managed to get motivated to do documentation before a release). To use these actions/modifiers, you'll need to get the SVN trunk version. Charles Jeff Wright [EMAIL PROTECTED] wrote on 08/17/2008 12:28:16 PM: Charles, Thanks for responding. That scenario does look like it will work, but I was certainly unaware of the existence of the strcmp and nop actions, as well as the condexec action modifier. I don't see these items in the documentation anywhere. I guess I could grep through the codebase and try to figure out all the sundry options available to me and how they work, but that was a bit more than I was originally hoping for. In any case, when I try this: recv response=200 optional=true next=1 action ereg regexp=.* search_in=hdr header=Contact: assign_to=contact/ strcmp assign_to=compareval variable=contact value= / test assign_to=empty variable=compareval value=0 test=equal / /action /recv nop condexec=empty action error message=Server header is present. /action /nop I get this result: strcmp 'assign_to' parameter, compareval is not a valid integer! BTW, my sipp version is: SIPp v2.0-TLS, version 20071128, built Jan 7 2008, 16:31:36 Any ideas? Jeffrey Wright System Test Engineering Manager Aztek Networks, Inc. -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Sat 8/16/2008 7:14 PM To: Jeff Wright Cc: Anonymous Incognito; sipp-users@lists.sourceforge.net; sipp- [EMAIL PROTECTED] Subject: Re: [Sipp-users] check_it My best suggestion would be to assign the captured value to a variable, something like (not 100% sure on syntax, but this should sketch the plan for you); then strcmp it to an empty string (returns 0 if equal), and test on the strcmp return. recv ereg assign_to=server search_in=header header=Server: regexp=.* / strcmp assign_to=compareval variable=server value= / test assign_to=empty variable=compareval value=0 test=equal / /recv nop condexec=empty action error message=Server header is present. /action /nop Charles Jeff Wright [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 08/16/2008 12:35 PM To Anonymous Incognito [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] check_it This is the exact same thing I need to do (see my post from a couple of days ago). Please let me know if you find out a way to do it. Jeffrey Wright System Test Engineering Manager Aztek Networks, Inc. -Original Message- From: [EMAIL PROTECTED] on behalf of Anonymous Incognito Sent: Sat 8/16/2008 6:50 AM To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] check_it Hi , I would like to write a scenario as below. Search the SIP message for the presence of a header, Server (for example). If it is present then I would like to fail the call. I am not able to achieve it using check_it. Cheers David - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] check_it
My best suggestion would be to assign the captured value to a variable, something like (not 100% sure on syntax, but this should sketch the plan for you); then strcmp it to an empty string (returns 0 if equal), and test on the strcmp return. recv ereg assign_to=server search_in=header header=Server: regexp=.* / strcmp assign_to=compareval variable=server value= / test assign_to=empty variable=compareval value=0 test=equal / /recv nop condexec=empty action error message=Server header is present. /action /nop Charles Jeff Wright [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 08/16/2008 12:35 PM To Anonymous Incognito [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] check_it This is the exact same thing I need to do (see my post from a couple of days ago). Please let me know if you find out a way to do it. Jeffrey Wright System Test Engineering Manager Aztek Networks, Inc. -Original Message- From: [EMAIL PROTECTED] on behalf of Anonymous Incognito Sent: Sat 8/16/2008 6:50 AM To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] check_it Hi , I would like to write a scenario as below. Search the SIP message for the presence of a header, Server (for example). If it is present then I would like to fail the call. I am not able to achieve it using check_it. Cheers David - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] problem in compiling SIPp using GSL
What Linux distribution are you using? On redhat based distributions you must have gsl-devel installed as well. Charles amar mahmoud [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 08/14/2008 04:32 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] problem in compiling SIPp using GSL Hey, I want to use different distributions, so I need to use GSL, I have installed it , but when trying to compile SIPp as following: 1) installing GSL 2) uncomment the lines in local.mk ( under SIPp directory) 3) make it gives me the following output: [EMAIL PROTECTED]:~/sipp.svn$ make make OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` sipp make[1]: Entering directory `/home/amar/sipp.svn' g++ -D__LINUX -pthread -DSVN_VERSION=\unknown\ -DHAVE_GSL -I`if test -f /usr/local/lib/libgsl.so; then echo /usr/local; else echo ./ext; fi;`/include -I. -I/usr/include/openssl -c -o message.o message.cpp In file included from scenario.hpp:32, from sipp.hpp:63, from message.cpp:38: stat.hpp:45:25: error: gsl/gsl_rng.h: No such file or directory stat.hpp:46:29: error: gsl/gsl_randist.h: No such file or directory stat.hpp:47:25: error: gsl/gsl_cdf.h: No such file or directory In file included from scenario.hpp:32, from sipp.hpp:63, from message.cpp:38: stat.hpp:629: error: ISO C++ forbids declaration of ?gsl_rng? with no type stat.hpp:629: error: expected ?;? before ?*? token stat.hpp:652: error: ISO C++ forbids declaration of ?gsl_rng? with no type stat.hpp:652: error: expected ?;? before ?*? token stat.hpp:665: error: ISO C++ forbids declaration of ?gsl_rng? with no type stat.hpp:665: error: expected ?;? before ?*? token stat.hpp:678: error: ISO C++ forbids declaration of ?gsl_rng? with no type stat.hpp:678: error: expected ?;? before ?*? token stat.hpp:691: error: ISO C++ forbids declaration of ?gsl_rng? with no type stat.hpp:691: error: expected ?;? before ?*? token stat.hpp:705: error: ISO C++ forbids declaration of ?gsl_rng? with no type stat.hpp:705: error: expected ?;? before ?*? token stat.hpp:718: error: ISO C++ forbids declaration of ?gsl_rng? with no type stat.hpp:718: error: expected ?;? before ?*? token can any one tell me what wrong I did. Thanks sipp-users@lists.sourceforge.net Amar Got Game? Win Prizes in the Windows Live Hotmail Mobile Summer Games Trivia Contest Find out how. - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] UDP CHECKSUM ERROR with own UAS scenario
You can get the part between using a regular expression. Charles michael [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/30/2008 10:31 AM To sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] UDP CHECKSUM ERROR with own UAS scenario Nevermind, problem solved. Pebkac, as usual. Although, the [last_from] directive returns the whole From address from the las message, often in format such as name sip:[EMAIL PROTECTED]:port. Is it possible to get just the actual address, ie sip:[EMAIL PROTECTED], so that it can be used for Requests, ie NOTIFY sip:[EMAIL PROTECTED]:port SIP/2.0? - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Force source IP source Port at IP laye
No. If you want to spoof addresses it would be pretty hard to implement. If you just want to pick from one of your IP injection files, you could implement it without major code changes. Charles Cyrille OLIVIER [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/30/2008 10:40 AM To Ricardo Fernandes [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] Force source IP source Port at IP laye Hi Ricardo all, Thanks a lot for your answer. I will get the lastest version in the trunk source code, Unfortunatelly, i need to set the *source* IP and port of my sipp messages. so it leads to the 100$ question: does the setsrc or setsource option also exists ? :) BR, Cyrille Date: Wed, 30 Jul 2008 15:31:57 +0100 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Sipp-users] Force source IP source Port at IP laye Hello, I think you had the same problem as me. You can change the place here your sipp messages go to by specifying the host and the port in the scenario file like this: nop action setdest host=112.12.12.12 port=5060 protocol=udp / /action nop This works in sipp with UDP, in TCP i don't know, never tried. This will only work with the lastest version in the trunk source code, the current binaries in the sourceforge stable version will not recognized the setdest keyword. Ps: If you are using windows get this version from the snapshots here: http://sipp.sourceforge.net/snapshots/sipp-win32-2008-07-18.exe Hope it helps Ricardo Fernandes On Wed, Jul 30, 2008 at 2:31 PM, Cyrille OLIVIER [EMAIL PROTECTED] wrote: Hi all, I worried a bit about this post ;) Actually, I don't need a very developped answer but just a short (but clear ;) ) one. Of course, if needed, i can detail more Best regards thanks a lot, Cyrille From: [EMAIL PROTECTED] To: sipp-users@lists.sourceforge.net Date: Thu, 10 Jul 2008 10:13:50 + Subject: [Sipp-users] Force source IP source Port at IP layer Dear sipp-users, Again, I asked my requests about SIPp client using TCP: Is it possible to force sipp to use specific IP source Port source, at IP layer, for send messages when TCP with single socket (option '-t t1' used) ? I tried many things: 1/ -bind_local: seems unuseful. 2/ -i x.x.x.x -p options: it's only for some SIP headers but not for IP packet header. 3/ send -source_ip=x.x.x.x -source_port= for INVITE message: does not seems to work. 4/ look for this subject in mailing list archives: some conversation are closed to my question but not similar at 100% ... Currently, I don't know which other workaround or things to do :( I would really appreciate any help about that Thanks a lot, BR, Cyrille Discutez gratuitement avec vos amis en vidéo ! Téléchargez Messenger, c'est gratuit ! Avec Windows Live Messenger restez en contact avec tous vos amis ! Téléchargez Messenger, c'est gratuit ! - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users Consultez vos emails sur votre mobile ! Créez un compte Hotmail, c'est gratuit ! - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Redirection functionality from single UAS scenario xml
If it is a byte-for-byte match I can't think of any way to get around that either with the same or separate calls. The only thing you might be able to do is terminate the call, and use infindex to record the fact that you've seen this invite before. I suspect you might need to introduce some new keywords/actions. Charles Evgeny Miloslavsky [EMAIL PROTECTED] 07/29/2008 09:45 AM To Charles P Wright/Watson/[EMAIL PROTECTED] cc Subject RE: [Sipp-users] Redirection functionality from single UAS scenario xml Because SIPp recognizes redirected invite as retransmission of the initial one Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 / 7320 -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 29, 2008 4:40 PM To: Evgeny Miloslavsky Cc: sipp-users@lists.sourceforge.net; [EMAIL PROTECTED] Subject: Re: [Sipp-users] Redirection functionality from single UAS scenario xml I didn't see your script, but why can't you handle it as a single call flow like? recv INVITE send 302 recv INVITE Do the normal call flow here. Charles Evgeny Miloslavsky [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/29/2008 08:01 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Redirection functionality from single UAS scenario xml HI I?m testing the redirection feature of my DUT and I need both SIPp-UAS?s (redirector and final UAS responder) to be run from a single xml scenario file. I prepared one, but the problem is that SIPp does not distinguishing between a redirected INVITE request and the initial one. I tried to use labels but it doesn?t work. Any advices? PS: script is attached. Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 / 7320 - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPP command not running
Your XML must have a [fieldN] keyword and is expecting an injection file. Charles [EMAIL PROTECTED] wrote on 07/29/2008 11:27:09 AM: Have anyone send this error msg ? [EMAIL PROTECTED]:~/sbc# sipp -sf uac_reg_sample.xml 10.88.225.187 - trace_msg -m 1 2008-07-29 11:19:25:9181217344765.918059: No injection file was specified! - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Variable in recv timeout
The timeout field does not accept a variable as input. Charles Ricardo Fernandes [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/24/2008 06:14 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Variable in recv timeout Hello, How can i put a variable in the timeout field of recv. Ex: nop action assignstr assign_to=MillisToWaitTemp value=[field2]/ todouble assign_to=MillisToWait variable=MillisToWaitTemp/ /action nop recv request=BYE timeout=[$MillisToWait] ontimeout=Label1 next=LabelBye/ This gives me an error: message timeout 'timeout' parameter, [$MillisToWait] is not a valid integer! Is there a way to make this work, or is this a illegal instruction? TIA Ricardo Fernandes - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPP INFO and REFER Method
SIPp doesn't actually pay attention to the methods; so it doesn't support or not support any of them. Charles Jad Haddad [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/21/2008 08:37 AM To Hadriel Kaplan [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject [Sipp-users] SIPP INFO and REFER Method Thank you Hadriel for replying But in fact, the [Domain] field, is a domain that 've added, and when we lunch the command sipp, all we have to do is to add the option -key [Domain] [Any Value We Want] in fact, the problem in my scenarios is that Sipp doesn't recognize the methods INFO or REFER. evrytime time i put these methos as comments, my scenarios work properly with the Invite method, but when i add these methods( REFER, INFO) SIPp does'nt recognze them, and i receive unsupported keyword. JAD From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Date: Sun, 20 Jul 2008 17:49:44 -0400 Subject: RE: [SIPForum-discussion] SIPP INFO and REFER Method Sipp questions should go to the sipp mailing list (sipp-users@lists.sourceforge.net), which you can subscribe to at: https://lists.sourceforge.net/lists/listinfo/sipp-users In sipp, anything inside brackets ?[]? in a scenario file is a keyword, and if sipp doesn?t recognize the word then it says unsupported keyword. Your particular problem is probably the ?[Domain]? keyword you have used in the messages, as I don?t see that in the keyword list for sipp 3.0. -hadriel From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jad Haddad Sent: Thursday, July 17, 2008 11:42 AM To: [EMAIL PROTECTED] Subject: [SIPForum-discussion] SIPP INFO and REFER Method Hello everyone I am testing a SIP interconnexion with a telecom operator using SIPP and i need to test the REFER and INFO method. I'm using SIPP as a UAC and i've tried many scenarios for these methods, i am always receiving the same error from SIPP '' Unsupported Keyword ' I would ask, if someone have already used such scenarios or have used SIPP and may helpe me resolving this problem. 10x a lot. Attached, you can find Scenarios i am trying for REFER and INFO method. Explore the seven wonders of the world Learn more! Connect to the next generation of MSN Messenger Get it now! - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp3.1 - scenario breaks whith exitcode 139
The dead calls should not keep SIPp open, but will be listed as tasks for 32 seconds. The dead calls help understanding the error logs, because otherwise they are listed as out-of-call messages, or even worse on a UAS as an unexpected message at index 0. The -m option should definitely work, I use it very often. Can you paste your command and a screen log? Charles [EMAIL PROTECTED] wrote on 07/14/2008 05:45:05 AM: Hi Charles, my testcases working fine with the latest SVN trunk but a other problem appears now. A scenario was be launched with the -m (n) option do not stop after n iterations. The scenario was be marked as a Dead call at the end, the following messages will be dicarded but the test never ends. A possible workaround for me is to define a explicit exit in any cases using the internal command stop_now. Wolfgang Deutsche Telekom AG Zentrum Technik Einführung Wolfgang Kanngießer Winterfeldtstraße 21, D-10781 Berlin -Ursprüngliche Nachricht- Von: Charles P Wright [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 9. Juli 2008 17:23 Wolfgang, I tried the scenario with the SVN trunk and did not get an error. Charles Kanngiesser, Wolfgang [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/07/2008 07:07 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] SIPp3.1 - scenario breaks whith exitcode 139 Hi all, recently I am asking for a error segmentation fault due to the external commands, i.e. action exec command=echo pass verdict.log/ /action The problem is still relevant but all scenarios worked properly until sipp2.0. There are any hints? Thanks,Wolfgang. Deutsche Telekom AG Zentrum Technik Einführung Wolfgang Kanngießer Winterfeldtstraße 21, D-10781 Berlin - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp for Automation
If the scenarios require precise synchronization between all the elements for a single call I would suggest using the extended third party call control (3pcc) feature of SIPp. Although its name is 3pcc, really it is just a mechanism to send data between SIPp instances. Charles Naresh [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/09/2008 01:08 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] SIPp for Automation hi, Can anyone suggest be any good framework or technique that shall help me to automate using SIPp Test Tool. I am currently having a basic framework using shell scripting and SIPp, but finding very difficult to synchronize the events. My requirement goes like this Scenario one uses UAC, UAS1, UAS2 Scenario two uses UAC, UAS1 Scenario three uses UAC, UAS1...UAS4 and so on I need to automate these scenarios. regards, Naresh Meet people who discuss and share your passions. Join them now. - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to force SIPPp to use the port specified in the scenario?
Tomasz, Just sending a note to confirm what you've written; you've got it all correct. The only minor thing is that in the latest trunks there is a setdest action that will let you change the host/port to send to. Charles Tomasz Radziszewski [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/09/2008 03:40 AM To sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] How to force SIPPp to use the port specified in the scenario? Hi, Unfortunately, as far as I know, sipp doesn't look at headers at all when selecting port (somebody please correct me if I'm wrong, because I don't have much experience with versions 1.1). Is the rrs=true in the recv scenario making a difference? (I actually never used routes in the scenario. I guess I may as well remove it). RRS is not for actually selecting port, but for storing Record-Route and Contect so that you can then use keywords [routes] and [next_url]. You should not remove RRS, because it manages not only Record-Route, but also Contact (and the contact is put to [next_url]). However, these keywords only affect message contents, and not the actual host/port it is sent to. If I use the command line -rsa host:port Yes, it will force (I use it in all or almost all of my tests). BTW, is SIPp using the port in Via header or what? Where does it gets the port to answer to from? I think it is just as you said - the port from where the request comes (unless -rsa is used). And since later in the scenario I need to send BYE to another proxy on port 5080, if I put -rsa localHost:5060 would that affect the BYE message too? Yes, RSA affects all messages What can I do to specify the ports to use in different part of the scenarios? I think this is impossible. When I needed such test, I used an additional proxy between sipp and the actual System-Under-Test. The proxy was sending the messages appropriately, based on headers. BR - Tomasz Radziszewski Senior Software Engineer Ericpol Telecom sp. z o.o. Madalinskiego 9, 30-303 Krakow, Poland e-mail: [EMAIL PROTECTED] http://www.ericpol.com/ - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp3.1 - scenario breaks whith exitcode 139
Wolfgang, I tried the scenario with the SVN trunk and did not get an error. Charles Kanngiesser, Wolfgang [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/07/2008 07:07 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] SIPp3.1 - scenario breaks whith exitcode 139 Hi all, recently I am asking for a error segmentation fault due to the external commands, i.e. action exec command=echo pass verdict.log/ /action The problem is still relevant but all scenarios worked properly until sipp2.0. There are any hints? Thanks,Wolfgang. Deutsche Telekom AG Zentrum Technik Einführung Wolfgang Kanngießer Winterfeldtstraße 21, D-10781 Berlin - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Patch for make on non-gnu systems
I've put this in the latest trunk. My mail reader or yours messed up the patch, if you can please send patches as MIME attachments. They are harder to read inline, but are easier to apply. Charles [EMAIL PROTECTED] wrote on 07/07/2008 06:18:27 PM: Hey guys, on my system make isn't GNU make so the make commands fail. By switching the use of make to $(MAKE) in the top level makefile I'm able to build your project. This shouldn't impact anything except for users where GNU make is installed as gnumake or gmake which will now be able to compile the project without changing the makefile. I've attached a patch for this. -Alfred Index: Makefile === --- Makefile (revision 494) +++ Makefile (working copy) @@ -161,43 +161,43 @@ # Building without TLS and authentication (no openssl pre-requisite) all: - make OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` $(OUTPUT) + $(MAKE) OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` $(OUTPUT) # Building with TLS and authentication ossl: - make OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_TLS=auth.o sslinit.o sslthreadsafe.o milenage.o rijndael.o TLS_LIBS=-lssl -lcrypto TLS=-D_USE_OPENSSL -DOPENSSL_NO_KRB5 $(OUTPUT) + $(MAKE) OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_TLS=auth.o sslinit.o sslthreadsafe.o milenage.o rijndael.o TLS_LIBS=-lssl - lcrypto TLS=-D_USE_OPENSSL -DOPENSSL_NO_KRB5 $(OUTPUT) #Building with PCAP play pcapplay: - make OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_PCAPPLAY=send_packets.o prepare_pcap.o PCAPPLAY_LIBS=-lpcap PCAPPLAY=-DPCAPPLAY $(OUTPUT) + $(MAKE) OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_PCAPPLAY=send_packets.o prepare_pcap.o PCAPPLAY_LIBS=-lpcap PCAPPLAY=-DPCAPPLAY $(OUTPUT) pcapplay_ossl: - make OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_TLS=auth.o sslinit.o sslthreadsafe.o milenage.o rijndael.o TLS_LIBS=-lssl -lcrypto TLS=-D_USE_OPENSSL -DOPENSSL_NO_KRB5 OBJ_PCAPPLAY=send_packets.o prepare_pcap.o PCAPPLAY_LIBS=-lpcap `if test -f ./ext; then echo - L./ext/lib; fi;` PCAPPLAY=-DPCAPPLAY `if test -f ./ext; then echo -I./ext/include; fi;` $(OUTPUT) + $(MAKE) OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_TLS=auth.o sslinit.o sslthreadsafe.o milenage.o rijndael.o TLS_LIBS=-lssl - lcrypto TLS=-D_USE_OPENSSL -DOPENSSL_NO_KRB5 OBJ_PCAPPLAY=send_packets.o prepare_pcap.o PCAPPLAY_LIBS=-lpcap `if test -f ./ext; then echo -L./ext/lib; fi;` PCAPPLAY=-DPCAPPLAY `if test -f ./ext; then echo -I./ext/include; fi;` $(OUTPUT) pcapplay_hp_li_ia: - @_HPUX_LI_FLAG=-D_HPUX_LI ; export _HPUX_LI_FLAG ; make pcapplay + @_HPUX_LI_FLAG=-D_HPUX_LI ; export _HPUX_LI_FLAG ; $(MAKE) pcapplay pcapplay_ossl_hp_li_ia: - @_HPUX_LI_FLAG=-D_HPUX_LI ; export _HPUX_LI_FLAG ; make pcapplay_ossl + @_HPUX_LI_FLAG=-D_HPUX_LI ; export _HPUX_LI_FLAG ; $(MAKE) pcapplay_ossl pcapplay_cygwin: - make OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_PCAPPLAY=send_packets.o prepare_pcap.o PCAPPLAY_LIBS=-lwpcap PCAPPLAY=-DPCAPPLAY $(OUTPUT) + $(MAKE) OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_PCAPPLAY=send_packets.o prepare_pcap.o PCAPPLAY_LIBS=-lwpcap PCAPPLAY=-DPCAPPLAY $(OUTPUT) pcapplay_ossl_cygwin: - make OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_TLS=auth.o sslinit.o sslthreadsafe.o milenage.o rijndael.o TLS_LIBS=-lssl -lcrypto TLS=-D_USE_OPENSSL -DOPENSSL_NO_KRB5 OBJ_PCAPPLAY=send_packets.o prepare_pcap.o PCAPPLAY_LIBS=-lwpcap PCAPPLAY=-DPCAPPLAY $(OUTPUT) + $(MAKE) OSNAME=`uname|sed -e s/CYGWIN.*/CYGWIN/` MODELNAME=`uname -m|sed s/Power Macintosh/ppc/` OBJ_TLS=auth.o sslinit.o sslthreadsafe.o milenage.o rijndael.o TLS_LIBS=-lssl - lcrypto TLS=-D_USE_OPENSSL -DOPENSSL_NO_KRB5 OBJ_PCAPPLAY=send_packets.o prepare_pcap.o PCAPPLAY_LIBS=-lwpcap PCAPPLAY=-DPCAPPLAY $(OUTPUT) $(OUTPUT): $(OBJ_TLS) $(OBJ_PCAPPLAY) $(OBJ) $(CCLINK) $(LFLAGS) $(MFLAGS) $(LIBDIR_$(SYSTEM)) \ $(DEBUG_FLAGS) -o $@ $(OBJ_TLS) $(OBJ_PCAPPLAY) $(OBJ) $(LIBS) $(TLS_LIBS) $(PCAPPLAY_LIBS) $(EXTRAENDLIBS) debug: - DEBUG_FLAGS=-g -pg ; export DEBUG_FLAGS ; make all + DEBUG_FLAGS=-g -pg ; export DEBUG_FLAGS ; $(MAKE) all debug_ossl: - @DEBUG_FLAGS=-g ; export DEBUG_FLAGS ; make ossl + @DEBUG_FLAGS=-g ; export DEBUG_FLAGS ; $(MAKE) ossl debug_pcap_cygwin: - @DEBUG_FLAGS=-g ; export DEBUG_FLAGS ; make pcapplay_ossl_cygwin +
Re: [Sipp-users] Again : Can Sipp have multiple remote_ip acting as a Sip Client ?
You can change the address things are sent to in the latest trunk using the setdest action. Something like setdest host=[$host] port=[$port] protocol=udp / You will need to combine it with regular expressions. In UAS mode SIPp responds to the address that sent it a message. You can also change the SIPp source code to do something like that for UAC mode as well. Charles [EMAIL PROTECTED] wrote on 07/08/2008 07:51:30 AM: Hello, I send this message to the forum last week, but i haven't obtain no anwser yet. Does anyone know if this is possible or not, i need to know as soon as possible, i am facing deadlines. Just need a anwser yes or no and if yes how to do it. Thanks in advance Ricardo Hello, I am facing a problem with the [remote_ip] keyword when using sipp as a sipp caller client. I have tree sip server machines(machine A,B and C). Machine A is a Load Balancing machine that distributes the sip messages to B or C. I set up the remote_ip has beeing the machine A. Sipp send the sip message Invite to machine A and the sip messages 100,180 ,486 or 200 are received from machine A. Then i put the sipp client on hold, but the Invite message to sipp cames from machine B, so the ack must go to machine B, but because i defined that the [remote_ip] is machine A sipp sends the request to machine A when sipp should send it to B. Is there a way to change the [remote_ ip] on run time on the scenario file? In the atachment goes my scenario file. In this url http://www.tech-invite.com/Ti-sip-service-1.html is a example of what i am trying to acomplish. Regards Ricardo Fernandes [attachment ScenarioHold.xml deleted by Charles P Wright/Watson/IBM] - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Again : Can Sipp have multiple remote_ip acting as a Sip Client ?
You will need to download and compile the subversion trunk version. Charles Ricardo Fernandes [EMAIL PROTECTED] 07/08/2008 10:13 AM To Charles P Wright/Watson/[EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net, [EMAIL PROTECTED] Subject Re: [Sipp-users] Again : Can Sipp have multiple remote_ip acting as a Sip Client ? Hello, I have installed the version sipp-win32-3.1.1.exe for windows and and i have changed the scenario file to recv request=BYE/recv nop action setdest host=172.21.29.2 port=5060 protocol=udp / /action /nop send ![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Contact: ricsip:[local_ip]:[local_port];transport=[transport] Content-Length: 0 ]] /send and an error occurs when i launch sipp 2008-07-08 15:01:15:3701215525675.370099: Unknown action: setdest. I have tried also with the unstable version sipp-win32-3.1.2(ossl) with the same result, must i set a switch in the command line for this to work. I would apreciate all the help you can give me. TIA Ricardo Fernandes On Tue, Jul 8, 2008 at 1:00 PM, Charles P Wright [EMAIL PROTECTED] wrote: You can change the address things are sent to in the latest trunk using the setdest action. Something like setdest host=[$host] port=[$port] protocol=udp / You will need to combine it with regular expressions. In UAS mode SIPp responds to the address that sent it a message. You can also change the SIPp source code to do something like that for UAC mode as well. Charles [EMAIL PROTECTED] wrote on 07/08/2008 07:51:30 AM: Hello, I send this message to the forum last week, but i haven't obtain no anwser yet. Does anyone know if this is possible or not, i need to know as soon as possible, i am facing deadlines. Just need a anwser yes or no and if yes how to do it. Thanks in advance Ricardo Hello, I am facing a problem with the [remote_ip] keyword when using sipp as a sipp caller client. I have tree sip server machines(machine A,B and C). Machine A is a Load Balancing machine that distributes the sip messages to B or C. I set up the remote_ip has beeing the machine A. Sipp send the sip message Invite to machine A and the sip messages 100,180 ,486 or 200 are received from machine A. Then i put the sipp client on hold, but the Invite message to sipp cames from machine B, so the ack must go to machine B, but because i defined that the [remote_ip] is machine A sipp sends the request to machine A when sipp should send it to B. Is there a way to change the [remote_ ip] on run time on the scenario file? In the atachment goes my scenario file. In this url http://www.tech-invite.com/Ti-sip-service-1.html is a example of what i am trying to acomplish. Regards Ricardo Fernandes [attachment ScenarioHold.xml deleted by Charles P Wright/Watson/IBM] - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to cut SIP-URI from Contact header
Tomasz and Evgeny, I have encountered this problem as well. I think the latest trunk should respect the quotes, but svn is down so I can't tell you what version is the minimum. Charles Tomasz Radziszewski [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/07/2008 06:03 AM To sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] How to cut SIP-URI from Contact header Hi Your expression contains and probably the XML parser reads it as closing of the ereg tag. It should, because it's within quotes, but from my experience it does. You can to replace the with HTML-like entity gt;, so ereg regexp= sip:[^;gt;]+ search_in=hdr header=Contact: assign_to=2 / BR Tomasz Radziszewski Senior Software Engineer Ericpol Telecom sp. z o.o. Madalinskiego 9, 30-303 Krakow, Poland e-mail: [EMAIL PROTECTED] http://www.ericpol.com/ Hi I have a following procedure to cut SIP-URI from Contact header of received message: action ereg regexp= sip:[^;]+ search_in=hdr header= Contact: assign_to=2 / /action Every time I try to run the scenario I get the assign_to value is missing message. Any advises? PS: I think that the problem is the char within the regex sip:[^;]+ Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 / 7320 http://www.juniper.net/ - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] AKAv1-MD5
You'll need a recent SIPp trunk version and to use the verifyauth action. It looks something like recv request=REGISTER / verifyauth assign_to=goodauth username=username password=password / /recv username and password can be any message substitution, and you can do branching based on the return value stored in goodauth. Charles Venkat Narasimhan [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/04/2008 08:05 AM To sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] AKAv1-MD5 Sorry for my previous incomplete mail Consider the following scenario ... in this scenario, how can i actually verify if the peer has sent the correct response in Authorization field? in the second REGISTER ? scenario name=AKAv1-MD5_BASIC recv request=REGISTER /recv send ![CDATA[ SIP/2.0 401 Unauthorized [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Max-Forwards: 70 WWW-Authenticate: Digest algorithm=AKAv1-MD5,nonce=dcd98b7102dd2f0e8b11d0f600bfb0c093,opaque=5ccc069c403ebaf9f0171e9517f40e41,qop=auth,auth-int,realm=localhost Content-Length: 0 ]] /send recv request=REGISTER action ereg check_it = true regexp=Authorization: ([[:alnum:]]*) search_in=msg assign_to=12/ ereg check_it = true regexp=response=([[:alnum:]]*) search_in=Authorization assign_to=resp/ /action /recv send retrans=500 ![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Contact: user sip:[EMAIL PROTECTED]:5060;expires=3600 Expires: 3600 Content-Length: 0 ]] /send /scenario Any help is appreciated Regards Venkat Not happy with your email address? Get the one you really want - millions of new email addresses available now at Yahoo! Not happy with your email address? Get the one you really want - millions of new email addresses available now at Yahoo! - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Variable $5 is referenced 1 times!
The only referenced once check is there to prevent typos and other similar errors that would have SIPp load a scenario with a variable only used once (the theory being why do you need to read or write to a variable if you never read or write from it again, basically like an unused variable warning from your compiler). If you don't need $5, you can use it for something unneeded like: assign assign_to=5 value=0 / I would rename 5 to something like dummy (you can use string names not just numeric names, which makes the scenario much more readable). A better long term solution would be to modify the SIPp source code to make the whole match variable optional so that you don't need to jump through hoops like this. Charles Sajith T S [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/04/2008 09:28 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Variable $5 is referenced 1 times! Hi, I'm trying to find the contact uri from a 200 OK response sent by the UAC, but sipp apparently isn't happy about the scenario file syntax. I keep getting this error: Variable $5 is referenced 1 times! is this correct? recv response=200 rtd=true rrs=true action ereg regexp=sip:.*;transport=.*SIP/2.0 search_in=hdr header=Contact: check_it=true assign_to=5,6 / /action /recv Thanks, Sajith. -- the lyf so short, the craft so long to lerne. -- Chaucer. - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] stat CSV log files mess up in Excel.
You can also use -stat_delimiter , on the command line. Charles Tu Le Van [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/01/2008 03:24 AM To 'Lucian Romi' [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] stat CSV log files mess up in Excel. Hi Romi, Try to edit by using wordpad and replace ; with , From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lucian Romi Sent: Tuesday, July 01, 2008 2:21 AM To: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] stat CSV log files mess up in Excel. Attachment is my output. Looks like something's wrong with the separator. On Mon, Jun 30, 2008 at 12:19 PM, Lucian Romi [EMAIL PROTECTED] wrote: Hi, all I tried to analysis statistic output in excel. However, it got mess up. Some colums went together. There maybe something wrong with the separator. Can you help me figure out this one? Thanks! - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] why i can't download the original xml file?
They don't seem to exist, but you can just do; ./sipp -sd [scenario] to print out one of the default scenarios. Charles Mike Li [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 07/01/2008 05:31 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] why i can't download the original xml file? Hi, Could some body tell me why i can't download the original xml file from http://sipp.sourceforge.net/doc3.0/reference.html ? or Who can send these files to me, I'm new to SIPp. TIA Mike - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to do REGISTER and UAS(INVITE server) together.
You have to patch the current source code, but it would involve significant reimplementation of the original patch. Charles Lucian Romi [EMAIL PROTECTED] wrote on 06/25/2008 07:12:53 PM: Hi, I google and figure out that somebody actually implemented something call pre-scenario post-scenario to deal with this problem. Please follow this link http://osdir.com/ml/telephony.sipp.user/2006-09/msg00070.html Even I got the latest code from svn, the pre pos scenario is not there. Is there any other way to implement this with latest build or I have to patch the source code? On Mon, Jun 23, 2008 at 11:36 AM, Lucian Romi [EMAIL PROTECTED] wrote: Thanks Itzik and Charles. Looks like I have to use two different scenarios. Question are can I run these two scenarios into one sipp process? If not, can my REGISTER sender and INVITE server use the same source port to send messages? I need to do this because I'm inside the NAT. Thanks! On Sun, Jun 22, 2008 at 9:17 PM, Itzik Harel [EMAIL PROTECTED] wrote: Charles If this is the case for Lucian, than you are correct about the need to use two separate scripts. I probably did not understood Lucian request properly. Regards, Itzik Harel. -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Sunday, June 22, 2008 9:52 PM To: Itzik Harel Cc: Lucian Romi; sipp-users@lists.sourceforge.net; [EMAIL PROTECTED] Subject: Re: [Sipp-users] How to do REGISTER and UAS(INVITE server) together. Itzik, Your scenario has a single UAS that will handle either the REGISTER or INVITE. I believe what Lucian wants to do is have a single scenario that sends the REGISTER every hour and listens for INVITES. The first is possible (you did it), the second is not (it must be broken into two separate scenarios). Charles [EMAIL PROTECTED] wrote on 06/21/2008 11:42:34 PM: I did something once that does not require 2 scenarios. Using labels, you can create different flow for Register and Invite within the same scenario. This will also support different call-id's for these methods. The script I have attached shows Register + handling Invite as a redirect server, but I think you can integrate the Register and labels portion into a basic UAS scenario. Good luck, Itzik. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles P Wright Sent: Sunday, June 22, 2008 12:18 AM To: Lucian Romi Cc: [EMAIL PROTECTED]; sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] How to do REGISTER and UAS(INVITE server) together. You need to use separate scenarios. Charles Lucian Romi [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 06/20/2008 07:06 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] How to do REGISTER and UAS(INVITE server) together. Hi, I tried to create one scenario like this. There are REGISTER and INVITE server. To make the server able to locate this UAS without expire, every 3600 second will send 1 REGISTER. INVITE traffic is continusly sending from UAC, say 1 per second. Because REGISTER and INVITE server have different frequency and Call-ID, anybody tell me how to do this scenario like this. Thanks! -- -- - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- -- - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users [attachment redirect_sim.xml deleted by Charles P Wright/Watson/IBM] - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php
Re: [Sipp-users] How to do REGISTER and UAS(INVITE server) together.
You need to use separate scenarios. Charles Lucian Romi [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 06/20/2008 07:06 PM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] How to do REGISTER and UAS(INVITE server) together. Hi, I tried to create one scenario like this. There are REGISTER and INVITE server. To make the server able to locate this UAS without expire, every 3600 second will send 1 REGISTER. INVITE traffic is continusly sending from UAC, say 1 per second. Because REGISTER and INVITE server have different frequency and Call-ID, anybody tell me how to do this scenario like this. Thanks! - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to set behavior for any received response except of 200 OK
You can not use a != for matching, and actually I just checked the code. Regular expressions are only valid for requests not responses. You can add a label named _unexp.main and then exit on any unexpected message. For example, this could go at the end of your scenario. nop next=end / label id=_unexp.main / nop action error message=Got something aside from a 200 / /action /nop label id=end / Charles Evgeny Miloslavsky [EMAIL PROTECTED] 06/17/2008 02:57 AM To Charles P Wright/Watson/[EMAIL PROTECTED] cc Subject RE: [Sipp-users] How to set behavior for any received response except of 200 OK HI and thanks for your response. I need the following functionality: if received response != 200 Ok then exit. I need my SIPp instance exit the scenario in case received response is not 200 OK. My intuition says that it should be something like recv response !=200 action exec int_cmd=stop_now/ but I?m not sure that this kind of syntax is applicable for SIPp Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 / 7320 -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Monday, June 16, 2008 6:54 PM To: Dhananjaya Reddy Eadala Cc: Evgeny Miloslavsky; sipp-users@lists.sourceforge.net; [EMAIL PROTECTED] Subject: Re: [Sipp-users] How to set behavior for any received response except of 200 OK SIPp won't exit but the call will fail. If you want SIPp to exit, you'll need to do a regular expression match and an action that includes something like error message=Got something other than 200 /. Charles Dhananjaya Reddy Eadala [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 06/16/2008 11:12 AM To Evgeny Miloslavsky [EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net Subject Re: [Sipp-users] How to set behavior for any received response except of 200 OK in scenario, set the following: recv response=200 /recv If sipp receives other than 200, then it will exit automatically. Dhana On 6/16/08, Evgeny Miloslavsky [EMAIL PROTECTED] wrote: Hi All How do I cause my SIPP instance to exit the scenario for every received response except of 200 OK. Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 / 7320 - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to set behavior for any received response except of 200 OK
SIPp won't exit but the call will fail. If you want SIPp to exit, you'll need to do a regular expression match and an action that includes something like error message=Got something other than 200 /. Charles Dhananjaya Reddy Eadala [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 06/16/2008 11:12 AM To Evgeny Miloslavsky [EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net Subject Re: [Sipp-users] How to set behavior for any received response except of 200 OK in scenario, set the following: recv response=200 /recv If sipp receives other than 200, then it will exit automatically. Dhana On 6/16/08, Evgeny Miloslavsky [EMAIL PROTECTED] wrote: Hi All How do I cause my SIPP instance to exit the scenario for every received response except of 200 OK. Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 / 7320 - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Multiple User-Agents w/ RTP
You can look at the exit codes, which I believe should give you a pass/fail result. Charles Gomtesh Jain [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 06/13/2008 03:23 AM To sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] Multiple User-Agents w/ RTP Hi All, I am trying to automate SIP test cases. Is there any way to know the result(Pass/Fail) of a particular test case. Please let me know if any of you know . Regards Gomtesh - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Multiple User-Agents w/ RTP
I don't think SIPp currently supports this, but RTP is done with raw sockets so the modifications would not be terribly difficult for someone to do. Charles [EMAIL PROTECTED] wrote on 06/09/2008 06:43:42 PM: Hello, I've been able to emulate multiple user agents creating and destroying sessions by using the -inf, -t, and -ip_field parameters. While this is useful, I also need to be able to send RTP data during those sessions from multiple IP addresses. I can't seem to figure out how to configure SIPp to do this! I'm using modified versions of the uac_pcap.xml and uas.xml scenarios posted in the documentation. Like I've said, I can create sessions with different source IP addresses (ie, different user agents) but I don't know how to get SIPp to send RTP data from those different IP addresses! Is this even possible with the current version of SIPp? -Thanks in advance - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Display Scenario Name
You can add a nop with a display attribute like: nop display=String to Display / At the top of the scenario. Charles Naresh [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 06/04/2008 07:45 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Display Scenario Name Hi, I am using the SIPp to run various scenarios on a series, my setup includes SIPp acting as UAC, UAS1, UAS2?. This setup is basically to test our B2B server. My question for SIPp users is that - Is it possible to display a known text on the SIPp scenario screen that shall help me to identify the scenario that is currently running? oI would like to view the Scenario name displayed on the Screen 1. Regards, Send free SMS to your Friends on Mobile from your Yahoo! Messenger. Download Now! http://messenger.yahoo.com/download.php - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://sourceforge.net/services/buy/index.php ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Problem with 3PCC Extended scenarios
Michael, My guess is that you are bumping into an open-call limit on the master. You can remove that with -l 0 and it will follow the correct rate for you. If that doesn't work (or if it does) let me know, and I'll try looking at your scenarios and scripts in more detail. Charles [EMAIL PROTECTED] wrote on 05/30/2008 01:54:25 PM: I have a fairly complicated set of 3PCC Extended scenario files. I have a master SIPp instance that talks to a Network SIP Server (whose only job is to determine which of a number of Premise SIP Servers to send a call to) and receives a ?302 Moved Permanently?. The master scenario then determines the appropriate Premise SIP Server endpoint address from the contact header in the 302 message and sends the information to the appropriate slave SIPp instance (Using SendCmd), each of which is connected to a different Premise Sip Server. When the master sends a message to the slave, it is then finished its work and can exit, but if it does exit, the slaves die. To remedy this I tried placing a recvCmd in the master and added a sendCmd in the slave that will signal the master when the call has completed. The problem is, that If we put a SendCmd in the master followed by a recvCmd so we will be notified when the slave?s call has completed, the call rate is very low (Essentially, 1 call goes to each slave and no more calls are generated until those calls have completed). By removing the recvCmd from the master and replacing it with a pause of 60 seconds (Call duration is a little less than 60 seconds), we see the calls processed at the rate we expect (10 calls / sec default or whatever we put in ?r and ?rp). A pause of only 1 second causes the same problem as no pause. Inserting a pause of appropriate duration does make the problem go away, but the problem with this is that we don?t know how long the calls will be processed by the slave. When no agents are available the calls are queued and could be in the queue for a while. This makes it difficult to predict how long to make the pause , and we don?t want to put in something ridiculously large. The question is ? why does adding a recvCmd in the master and a sendCmd in the slave cause this behaviour? I have attached the shell scripts, the 3PCC Extended config files and the scenario files. They are called in the following order: callGeneratorToPremiseSIPServer-1.sh callGeneratorToPremiseSIPServer-2.sh callGeneratorToNetworkSIPServer.sh thanks for any help you can give, Michael Lynch CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain confidential and proprietary information of Alcatel-Lucent and/or its affiliated entities. Access by the intended recipient only is authorized. Any liability arising from any party acting, or refraining from acting, on any information contained in this e-mail is hereby excluded. If you are not the intended recipient, please notify the sender immediately, destroy the original transmission and its attachments and do not disclose the contents to any other person, use it for any purpose, or store or copy the information in any medium. Copyright in this e-mail and any attachments belongs to Alcatel-Lucent and/or its affiliated entities.[attachment callGeneratorToPremiseSIPServer-2.sh deleted by Charles P Wright/Watson/IBM] [attachment slave.cfg deleted by Charles P Wright/Watson/IBM] [attachment slave-1.cfg deleted by Charles P Wright/Watson/IBM] [attachment slave-2.cfg deleted by Charles P Wright/Watson/IBM] [attachment callGeneratorToNetworkSIPServer.sh deleted by Charles P Wright/Watson/IBM] [attachment callGeneratorToNetworkSIPServer.xml deleted by Charles P Wright/Watson/IBM] [attachment callGeneratorToPremiseSIPServer.xml deleted by Charles P Wright/Watson/IBM] [attachment callGeneratorToPremiseSIPServer-1.sh deleted by Charles P Wright/Watson/IBM] - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Double free or corruption in sipp
Can you repliate the bug? If so, please compile SIPp with -g support and provide the same backtrace. Also, if it is something simple to replicate with only SIPp please post the scenarios required to replicate it. Charles Gomtesh Jain [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/28/2008 03:42 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Double free or corruption in sipp Hi All, Can anyone explain me about this problem . -- Sipp Server Mode --- *** glibc detected *** ./sipp: double free or corruption (fasttop): 0x086b6580 * ** === Backtrace: = /lib/libc.so.6[0x49426a96] /lib/libc.so.6(cfree+0x90)[0x49429fb0] /usr/lib/libstdc++.so.6(_ZdlPv+0x21)[0x4a031691] /usr/lib/libstdc++.so.6(_ZdaPv+0x1d)[0x4a0316ed] ./sipp[0x808659e] ./sipp[0x806383c] ./sipp[0x8063e50] ./sipp[0x80672bb] ./sipp[0x80665b2] ./sipp[0x8078acc] ./sipp[0x807bb8f] /lib/libc.so.6(__libc_start_main+0xdc)[0x493d5dec] ./sipp(__gxx_personality_v0+0x371)[0x804bdf1] === Memory map: 005f4000-00623000 r-xp fd:00 9978382 /usr/local/lib/libgslcblas.so.0 .0.0 00623000-00624000 rwxp 0002e000 fd:00 9978382 /usr/local/lib/libgslcblas.so.0 .0.0 00b1b000-00c9b000 r-xp fd:00 9983619 /usr/local/lib/libgsl.so.0.10.0 00c9b000-00ca5000 rwxp 0017f000 fd:00 9983619 /usr/local/lib/libgsl.so.0.10.0 00fee000-00fef000 r-xp 00fee000 00:00 0 [vdso] 08048000-080c1000 r-xp fd:00 15044431 /root/sipp-3.0.src/sipp 080c1000-080c5000 rwxp 00078000 fd:00 15044431 /root/sipp-3.0.src/sipp 080c5000-0821b000 rwxp 080c5000 00:00 0 08652000-087d1000 rwxp 08652000 00:00 0 489f1000-48a0a000 r-xp fd:00 10847820 /lib/ld-2.5.so 48a0a000-48a0b000 r-xp 00019000 fd:00 10847820 /lib/ld-2.5.so 48a0b000-48a0c000 rwxp 0001a000 fd:00 10847820 /lib/ld-2.5.so 48a93000-48aa2000 r-xp fd:00 10847856 /lib/libresolv-2.5.so 48aa2000-48aa3000 r-xp e000 fd:00 10847856 /lib/libresolv-2.5.so 48aa3000-48aa4000 rwxp f000 fd:00 10847856 /lib/libresolv-2.5.so 48aa4000-48aa6000 rwxp 48aa4000 00:00 0 493c-494fa000 r-xp fd:00 10847845 /lib/libc-2.5.so 494fa000-494fc000 r-xp 0013a000 fd:00 10847845 /lib/libc-2.5.so 494fc000-494fd000 rwxp 0013c000 fd:00 10847845 /lib/libc-2.5.so 494fd000-4950 rwxp 494fd000 00:00 0 49502000-49527000 r-xp fd:00 10847847 /lib/libm-2.5.so 49527000-49528000 r-xp 00024000 fd:00 10847847 /lib/libm-2.5.so 49528000-49529000 rwxp 00025000 fd:00 10847847 /lib/libm-2.5.so 4952b000-4952d000 r-xp fd:00 10847846 /lib/libdl-2.5.so 4952d000-4952e000 r-xp 1000 fd:00 10847846 /lib/libdl-2.5.so 4952e000-4952f000 rwxp 2000 fd:00 10847846 /lib/libdl-2.5.so 49531000-49544000 r-xp fd:00 10847848 /lib/libpthread-2.5.so 49544000-49545000 r-xp 00012000 fd:00 10847848 /lib/libpthread-2.5.so 49545000-49546000 rwxp 00013000 fd:00 10847848 /lib/libpthread-2.5.so 49546000-49548000 rwxp 49546000 00:00 0 4954a000-4955c000 r-xp fd:00 9961478/usr/lib/libz.so.1.2.3 4955c000-4955d000 rwxp 00011000 fd:00 9961478/usr/lib/libz.so.1.2.3 49f7-49f7b000 r-xp fd:00 10847865 /lib/ libgcc_s-4.1.2-20070626.so .1 49f7b000-49f7c000 rwxp a000 fd:00 10847865 /lib/ libgcc_s-4.1.2-20070626.so .1 49f7e000-4a05e000 r-xp fd:00 9978620 /usr/lib/libstdc++.so.6.0.8 4a05e000-4a062000 r-xp 000df000 fd:00 9978620 /usr/lib/libstdc++.so.6.0.8 4a062000-4a063000 rwxp 000e3000 fd:00 9978620 /usr/lib/libstdc++.so.6.0.8 4a063000-4a069000 rwxp 4a063000 00:00 0 4bb54000-4bb94000 r-xp fd:00 9983588 /usr/local/lib/libcurses.so 4bb94000-4bb9c000 rwxp 0004 fd:00 9983588 /usr/local/lib/libcurses.so 4bb9c000-4bb9d000 rwxp 4bb9c000 00:00 0 4f17a000-4f297000 r-xp fd:00 10846428 /lib/libcrypto.so.0.9.8b 4f297000-4f2a9000 rwxp 0011d000 fd:00 10846428 /lib/libcrypto.so.0.9.8b 4f2a9000-4f2ad000 rwxp 4f2a9000 00:00 0 4f3e5000-4f3e7000 r-xp fd:00 10846426 /lib/libcom_err.so.2.1 4f3e7000-4f3e8000 rwxp 1000 fd:00 10846426 /lib/libcom_err.so.2.1 4f3ea000-4f47 r-xp fd:00 9971285/usr/lib/libkrb5.so.3.2 4f47-4f472000 rwxp 00086000 fd:00 9971285/usr/lib/libkrb5.so.3.2 4f474000-4f47b000 r-xp fd:00 9967703 /usr/lib/libkrb5support.so.0.1 4f47b000-4f47c000 rwxp 6000 fd:00 9967703 /usr/lib/libkrb5support.so.0.1 4f47e000-4f4a3000 r-xp fd:00 9970876 /usr/lib/libk5crypto.so.3.0 - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] (no subject)
For media support pcap is certainly required. I don't know what libnet is. I don't know where to get them for Solaris anymore, but sunfreeware.com may be a start. It used to be the place to get all the extra packages that you needed. Charles Monica Sam -X (monsam - WIPRO at Cisco) [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/28/2008 06:42 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] (no subject) Hi , I have a Solaris server : SunOS sbc-auto5 5.8 Generic_117350-34 sun4u sparc SUNW,Ultra-5_10 I do not have libnet or libpcap packages installed: pkginfo | grep *libnet* No match Has anyone installed SIPp with media support successfully on Solaris?If so, can you please guide me on how to install SIPp on Solaris with pcap support.Are these packages(libnet and libpcap) required?.Can you point me to the site from which I can dowload theses packages? Thanks in advance, Monica. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Sending requests from inital UAS
There shouldn't be any magic to it. Just copy the request from the UAC scenario to the UAS and the recv from the UAS to the UAC. Charles [EMAIL PROTECTED] wrote on 05/28/2008 05:59:22 AM: Hi, I'm trying to write a scenario where sipp should act as B side and I want to send BYE from sipp A - invite - sipp A - 100 trying - sipp A - 180 Ringing - sipp A - 200 OK - sipp A - ACK - sipp A - BYE - sipp A - 200 OK - sipp However I'm stuck with getting the correct To/From headers ( and requestln) when creating the BYE request. Anyone that know of an example where the initial uas side should act as auc and send a request? Attached are my scenario sofar ( call setup but no BYE from B) Regards, // Andreas [attachment b_scenario.sipp.xml deleted by Charles P Wright/Watson/IBM] - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Sending requests from inital UAS
I don't have any examples of that, but you should be able to get all of that with regular expressions. Charles Andreas Byström (Polystar T M) [EMAIL PROTECTED] 05/28/2008 08:20 AM To Charles P Wright/Watson/[EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net sipp-users@lists.sourceforge.net, [EMAIL PROTECTED] [EMAIL PROTECTED] Subject SV: [Sipp-users] Sending requests from inital UAS Doesnt the outgoing request have to be built with the following info (in my case, where I want the sipp script to be uas and create a BYE request and my sipp script has UAS when the dialog is set up): * Requestline - Should be whatever is in the Contact header in the request that started the dialog (unless there are some Route headers which it is not in my case) * To header - should be the same as the incoming From header * From header - should be the same as incoming To header + the tag created by B when sending responses to the first request If I use for example the send bye from the uac example, the requestln contians [service] and to/form has [local_ip]/[remote_ip]. That wont work since that is not how the incoming request that creates the dialog looks like. The incoming request to sipp script looks like this: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: sip:[EMAIL PROTECTED];tag=1a3da3c6 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 10.10.8.122:5060;branch=z9hG4bK87cda52f07c481e5a8778b4ded75 Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Content-Type: application/sdp Content-Length: 177 sdp not shown I'm guessing that I have the same problem creating this BYE as I would have had if I wanted the B side to create a re-Invite. Maybe there are some examples on creating reInvites from B side? Regards, // Andreas -Ursprungligt meddelande- Från: Charles P Wright [mailto:[EMAIL PROTECTED] Skickat: den 28 maj 2008 14:07 Till: Andreas Byström (Polystar T M) Kopia: sipp-users@lists.sourceforge.net; [EMAIL PROTECTED] Ämne: Re: [Sipp-users] Sending requests from inital UAS There shouldn't be any magic to it. Just copy the request from the UAC scenario to the UAS and the recv from the UAS to the UAC. Charles [EMAIL PROTECTED] wrote on 05/28/2008 05:59:22 AM: Hi, I'm trying to write a scenario where sipp should act as B side and I want to send BYE from sipp A - invite - sipp A - 100 trying - sipp A - 180 Ringing - sipp A - 200 OK - sipp A - ACK - sipp A - BYE - sipp A - 200 OK - sipp However I'm stuck with getting the correct To/From headers ( and requestln) when creating the BYE request. Anyone that know of an example where the initial uas side should act as auc and send a request? Attached are my scenario sofar ( call setup but no BYE from B) Regards, // Andreas [attachment b_scenario.sipp.xml deleted by Charles P Wright/Watson/IBM] - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Registration Issue
No, but the best way to try debugging it with such a small scale test would be to do -trace_msg so you can examine every message in the supposedly failed call to see if you see what is wrong. Which log is thorwing the errors on you? Charles Martin Ostrovsky [EMAIL PROTECTED] 05/28/2008 10:23 AM To Charles P Wright/Watson/[EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net, [EMAIL PROTECTED] Subject Re: [Sipp-users] Registration Issue Charles, thanks for your answer-question, In fact, i don't want any failure, but i read in the log some registration failures but in the final report i don't see any. do u have any idea? cheers, martin. On Wed, May 28, 2008 at 10:52 AM, Charles P Wright [EMAIL PROTECTED] wrote: Martin, Why are you expecting failures? Charles Martin Ostrovsky [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/28/2008 09:17 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Registration Issue Hi all, I'd need your help to understand this problem I am having: I run this command: ./sipp -sn uac 192.168.4.11 -i 192.168.4.78 -sf /home/mostro/tests/register_client.xml -inf /home/mostro/tests/register_client.csv -r 1 -m 1000 -l 1 -trace_err -d 1000 -auth_uri 192.168.4.11 where x.x.4.11 is the server ip and x.x.4.78 is my local ip. register_client.xml is the scenario where I try to register a phone and register_client.csv is a list of users. The problem is the following, when I finish the test, the report does not show any fail tests. That's strange. I've copied a part of the final report. Current Time | 2008-05-2809:15:22:7501211976922.750478 - +---+-- Counter Name | Periodic value| Cumulative value - +---+-- Elapsed Time | 00:00:00:758 | 00:00:06:762 Call Rate 1.319 cps |0.887 cps - +---+-- Incoming call created |0 |0 OutGoing call created |1 |6 Total Call created | |6 Current Call |0 | -+---+-- Generic counter 1 |1 |6 -+---+-- Successful call|1 |6 Failed call|0 |0 -+---+-- Call Length| 00:00:00:002 | 00:00:00:002 -- Test Terminated Thanks, Cheers. Martin. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Registration Issue
Martin, Why are you expecting failures? Charles Martin Ostrovsky [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/28/2008 09:17 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Registration Issue Hi all, I'd need your help to understand this problem I am having: I run this command: ./sipp -sn uac 192.168.4.11 -i 192.168.4.78 -sf /home/mostro/tests/register_client.xml -inf /home/mostro/tests/register_client.csv -r 1 -m 1000 -l 1 -trace_err -d 1000 -auth_uri 192.168.4.11 where x.x.4.11 is the server ip and x.x.4.78 is my local ip. register_client.xml is the scenario where I try to register a phone and register_client.csv is a list of users. The problem is the following, when I finish the test, the report does not show any fail tests. That's strange. I've copied a part of the final report. Current Time | 2008-05-2809:15:22:7501211976922.750478 - +---+-- Counter Name | Periodic value| Cumulative value - +---+-- Elapsed Time | 00:00:00:758 | 00:00:06:762 Call Rate 1.319 cps |0.887 cps - +---+-- Incoming call created |0 |0 OutGoing call created |1 |6 Total Call created | |6 Current Call |0 | -+---+-- Generic counter 1 |1 |6 -+---+-- Successful call|1 |6 Failed call|0 |0 -+---+-- Call Length| 00:00:00:002 | 00:00:00:002 -- Test Terminated Thanks, Cheers. Martin. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users register_client.xml Description: Binary data register_client.csv Description: Binary data - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] 3.1 - scenarios don't work, no documentation, not packaged properly
SVN is subversion (like CVS), which is the source code control that we use for SIPp development. With a new Fedora or RedHat you can probably type svn to see if you have it. If not you can use yum to install subversion. This page has the URL to the SIPp repository http://sipp.sourceforge.net/wiki/index.php/Dev, which is: https://sipp.svn.sourceforge.net/svnroot/sipp/sipp/trunk And this is a basic tutorial if you are familiar with CVS: http://svn.collab.net/repos/svn/trunk/doc/user/cvs-crossover-guide.html Charles Michael Lynch [EMAIL PROTECTED] 05/23/2008 10:23 AM To Charles P Wright/Watson/[EMAIL PROTECTED] cc Srivastava, Anuj Kumar [EMAIL PROTECTED], sipp-users@lists.sourceforge.net, [EMAIL PROTECTED] Subject RE: [Sipp-users] 3.1 - scenarios don't work, no documentation, not packaged properly Thanks Charles, Forgive my ignorance - what is SVN? And where can I get hold of the package containing it? Thanks, Michael -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Thursday, May 22, 2008 3:39 PM To: Michael Lynch Cc: Srivastava, Anuj Kumar; sipp-users@lists.sourceforge.net; [EMAIL PROTECTED] Subject: RE: [Sipp-users] 3.1 - scenarios don't work, no documentation, not packaged properly This is a bug that has been fixed in SVN. Charles Michael Lynch [EMAIL PROTECTED] 05/22/2008 12:58 PM To Charles P Wright/Watson/[EMAIL PROTECTED] cc Srivastava, Anuj Kumar [EMAIL PROTECTED], sipp-users@lists.sourceforge.net, [EMAIL PROTECTED] Subject RE: [Sipp-users] 3.1 - scenarios don't work, no documentation, not packaged properly Thanks Charles, We've got rid of some of the errors now. These are the latest versions of the scenario files and config files. They work in 3.0 and not in 3.1. The error we get from transfer-to-ss is:- In pcap pcap/dtmf_2833_1.pcap, npkts 10 max pkt length 24 base port 1 In pcap pcap/t30sec.pcap, npkts 1094 max pkt length 180 base port 49172 2008-05-21 10:13:10:0761211375590.076638: The label 'master' was not defined (index 9, ontimeout attribute) . (Also - XMLPad still picks up an error in the DTD shipped with both versions, so you might want to see if you can fix that one too.) Thanks, Michael -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Thursday, May 22, 2008 7:58 AM To: Michael Lynch Cc: Srivastava, Anuj Kumar; sipp-users@lists.sourceforge.net; [EMAIL PROTECTED] Subject: Re: [Sipp-users] 3.1 - scenarios don't work, no documentation, not packaged properly 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 loaded you should post it if at all possible. Charles Michael Lynch [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/21/2008 03:09 PM To Srivastava, Anuj Kumar [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] 3.1 - scenarios don't work, no documentation, not packaged properly Hi, Ok, done a bit more investigation and I?d like to share our progress with you and how we went about it ? because it was a bit arduous and hopefully someone will get some benefit from our experience even though we still can?t get it working with version 3.1. When we loaded the scenario into a different XML editor we found a missing tag. However ? we also found a problem in the DTD which might explain why our original XML editor didn?t pick up our missing tag. The error in the DTD we get is at line 21: !ATTLIST exec command CDATA - we don?t know what this is about or whether it is serious or not, anyway we fixed the missing tag in our XML, but we were then back to where we were before (almost), with the scenario working in 3.0 but not in 3.1. When we ran it in 3.0 it created calls and it worked. In 3.1 we got an error, it complained about the label ?m? which we used to denote the master. The way we successfully debugged was to remove the block with ?m? in it. SIPP then complained about something in the block above it, we deleted this block and guess what ? it complained about the block above that. We did this iteratively until SIPP started successfully and then we examined the block we had just removed. We found a couple of extraneous bits in there, an assignment which was never used and a tretrans on a receive command which obviously wasn?t necessary. During the debug process we found out that it didn?t like a timeout value of 180, having removed that it now works on v3.0 and performs very well ? still no joy with 3.1 though. Question ? is there a limit on values for timeouts? We need a long timeout because we have queuing on our sip server. Thanks, Michael From: Srivastava, Anuj Kumar [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 21, 2008 2:45 AM
