[Sipp-users] CSV Separator
The default SIPp CSV separator is not in fact a comma, but rather a semi-colon. This means that if you try to load the file directly in Excel you don't get an import wizard; and the data is unusable. Does anyone have objections to changing the default separator to a comma after the 3.0 release? Charles - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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] Record-Route Problem
Instructions for accessing the latest subversion source are here: http://sourceforge.net/svn/?group_id=104305 and here http://sipp.sourceforge.net/wiki/index.php/Dev. Charles Pradeep Mohapatra [EMAIL PROTECTED] wrote on 09/27/2007 06:20:29 AM: Hi Charles, Thanks for your quick response. Please let us know which latest version of SIPp we can use for Sun_OS-5.10. Regards, Pradeep -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 26, 2007 8:02 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]; SIPp Subject: Re: [Sipp-users] Record-Route Problem Santosh, What version of SIPp are you using? If you are not using the latest trunk, please upgrade and verify if the bug exists in that version. Charles [EMAIL PROTECTED] wrote on 09/26/2007 09:27:18 AM: Hi All, We are facing a problem in Record-Route While sending the Route header, the sipp is removing some closing from the Record-Route header and also sending ter instead of term somehow it is truncating last character from Record-Route. Any help on this is highly appreciated The script looks like this = recv response=200 rrs=true ontimeout=2 /recv and in ACK it has the [routes] keyword = The following is the trace = SIP/2.0 200 OK^M Via: SIP/2.0/UDP 10.106.5.86:7053;psrrposn=1; branch=z9hG4bK-6602-1-0;rport=7053^ M Record-Route: sip:10.106.5.166:5060;lr;term^M Record-Route: sip:[EMAIL PROTECTED]:5060;lr;term^M Record-Route: sip:[EMAIL PROTECTED]:5060;lr;term^M Record-Route: sip:[EMAIL PROTECTED]:5060;lr^M Record-Route: sip:[EMAIL PROTECTED]:5060;lr^M Record-Route: sip:10.106.5.166:5060;lr;orig^M From: RG_USER_18_1_2_2500 sip:[EMAIL PROTECTED];tag=1^M To: sip:[EMAIL PROTECTED];tag=34914^M Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE^M Contact: sip:10.106.5.223:5060^M Content-Type: application/sdp^M Content-Length: 137^M Supported: replace^M Supported: timer^M Session-Expires: 400;refresher=uas^M Server: SJphone/1.65.366d (SJ Labs)^M ^M v=0^M o=user1 53655765 2353687637 IN IP4 10.106.5.223^M s=-^M c=IN IP4 10.106.5.223^M t=0 0^M m=audio 1600 RTP/AVP 0^M a=rtpmap:0 PCMU/8000^M ^M ACK sip:10.106.5.223:5060 SIP/2.0^M Via: SIP/2.0/UDP 10.106.5.86:7053;psrrposn=1; branch=z9hG4bK-6602-1-0;rport=7053^ M From: RG_USER_18_1_2_2500 sip:[EMAIL PROTECTED];tag=1^M To: sip:[EMAIL PROTECTED];tag=34914^M Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK^M Contact: sip:[EMAIL PROTECTED]^M Max-Forwards: 70^M User-Agent: SJphone/1.65.362c (SJ Labs)^M P-Preferred-Identity: RG_USER_18_1_2_2500 sip: [EMAIL PROTECTED] .com^M Content-Length: 0^M Route: sip:10.106.5.166:5060;lr;orig, sip:[EMAIL PROTECTED] 106.5.167:506 0;lr, sip:[EMAIL PROTECTED]:5060;lr, sip: [EMAIL PROTECTED] .5.167:5060;lr;ter, sip:[EMAIL PROTECTED]:5060;lr;ter ,sip:10.10 6.5.166:5060;lr;ter^M = -- Thanks Regards, Santosh Reddy. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 DISCLAIMER == This e-mail may contain privileged and confidential information which is the property of Persistent Systems Pvt. 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 Pvt. Ltd. does not accept any liability for virus infected mails. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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] SIPP XML files for PUBLISH and SUBSCRIBE
Santosh, If you have the time, you should consider posting these to the Scenarios page of the Wiki at http://sipp.sourceforge.net/wiki/index.php/Scenarios. Charles [EMAIL PROTECTED] wrote on 09/27/2007 01:58:39 AM: Hi Bharath, I have attached two xml files, you can use these. Hope this helps you. On 9/27/07, Bharath Mundlapudi [EMAIL PROTECTED] wrote: Hi, I am looking for some sample XML scenario files for PUBLISH and SUBSCRIBE messages in sipp. Where can i find this information? Thanks in anticipation, Bharath - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 -- Thanks Regards, Santosh Reddy. Veraz - QA Persistent Systems Pvt Ltd. http://www.persistentsys.com [EMAIL PROTECTED] Mobile: +91-9890881924 Tel: +91-20 25678900 Extn: 2344 Dir: +91-20 25702344 [attachment publish.xml deleted by Charles P Wright/Watson/IBM] [attachment subscribe_notify.xml deleted by Charles P Wright/Watson/IBM] - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://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 2005. 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] Is it possible to launch multiple sipp instance onone PC?
There are two things that come to mind here, one that you can probably get some reasonable information now; but will miss things like unexpected messages. One thing that would make this nicer is if there were key words like [rtd number=2], etc. The best way may be to use the error or short message log and then parse it. Alternatively, it might make sense for stat.cpp to have a detailed log mode so that every time a counter is incremented, etc. there is a printout with the call id (which could then be correlated back to users using a log file. Charles Simon Flannery [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 09/27/2007 08:03 AM To Olav Kvittem [EMAIL PROTECTED], sipp-users sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] Is it possible to launch multiple sipp instance onone PC? Hi Olav, This may be way-off, but would loging a custom message to a file help? For example: recv request=INVITE crlf=true rrs=true action ereg regexp=.* search_in=hdr header=Some-New-Header: assign_to=1 / log message=From is [last_From]. Custom header is [$1]/ /action /recv or recv request=INVITE action exec command=echo [last_From] is the from header received from_list.log/ /action /recv Simon On 9/27/07, Olav Kvittem [EMAIL PROTECTED] wrote: Hello simon, [EMAIL PROTECTED] said: You should only need 1 or 2 SIPp instances for the UAC and UAS. A single SIPp instance can handle many users by using a CSV injection file. Just put ALL users in the one CSV file, not just BOB! I tried that, but discovered that the .csv report files did not contain the destination id's. I am trying to send repeated calls to different echoing proxys to make statistics so I need to know individual numbers. Is there a place I can hack to accomplish that ? And can i make the report files land in a different directory rather than that of the config file directory (patch sent to the list a couple of months ago). Olav - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 2005. 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] REGISTER before recv INVITE in one XML scenario
[EMAIL PROTECTED] wrote on 09/27/2007 08:45:12 AM: Registration works fine. And the INVITE package will send with the right Port to the UAS. But Sipp discard this massage because the CallID dosn't match (reference.html#Unexpected+messages) but i Need to Reset the callID because Register and recv always will have different one! Is there a solution. I didn't find any in doc or google. If i split the XML scenario file from sipp1 in to two parts, first to Register and the second to receive it works but this is a bad solution i think. Someone knows a solution? Splitting the register and UAS is actually the preferred solution. Charles - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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] Dual-interface support
You can use the -i option. Charles [EMAIL PROTECTED] wrote on 09/27/2007 05:08:52 PM: Is there a way with sipp to specify the interface from which the call is generated? I have a dual interface machine on different networks and believe the attempted call is heading out the wrong interface. Any input would be appreciated. Thank you. Got a little couch potato? Check out fun summer activities for kids. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 2005. 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] sipp as asterisk pbx extensions
Karthik, Yes, you can test Asterisk use SIPp in UAS mode as an extension. Charles [EMAIL PROTECTED] wrote on 09/26/2007 06:55:22 AM: Hi All, Can sipp tool be configured to acts a sip extensions such that to eliminate the need of sip softphone which acts as sip entities at different machines to ASterisk PBX. Please help Regards Karthik Caution -Disclaimer --- The information contained in the electronic message and any attachments to this message are intended for the exclusive use of the addressee(s)and may contain confidential or privileged information. If you are not intended recipient, please notify the sender immediately and destroy all copies of this message and any attachments. Computer viruses can be transmitted via email. The recipient advised to check this email and any attachments for the presence of viruses. Dexterity has taken responsible protection to prevent this risk and accepts no liability for any damage caused by any virus transmitted by this mail.[attachment winmail.dat deleted by Charles P Wright/Watson/IBM] - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 2005. 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] sipp as asterisk pbx extensions
Yes, you can run multiple instances of SIPp on one system. You just need to use different SIP and Media ports via the -p and -mp options, respectively. Charles Arumugam, Karthik [Dexterity] [EMAIL PROTECTED] wrote on 09/26/2007 10:13:25 AM: Hi Charles Thanks for your reply.. Can I run multiple instances of sip client acting as asterisk extensions from a single PC? Regards Karthik.Ajavascript:SetCmd(cmdSend); -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Wed 9/26/2007 6:38 PM To: Arumugam, Karthik [Dexterity] Cc: sipp-users@lists.sourceforge.net; [EMAIL PROTECTED] Subject: Re: [Sipp-users] sipp as asterisk pbx extensions Karthik, Yes, you can test Asterisk use SIPp in UAS mode as an extension. Charles [EMAIL PROTECTED] wrote on 09/26/2007 06:55:22 AM: Hi All, Can sipp tool be configured to acts a sip extensions such that to eliminate the need of sip softphone which acts as sip entities at different machines to ASterisk PBX. Please help Regards Karthik Caution -Disclaimer --- The information contained in the electronic message and any attachments to this message are intended for the exclusive use of the addressee(s)and may contain confidential or privileged information. If you are not intended recipient, please notify the sender immediately and destroy all copies of this message and any attachments. Computer viruses can be transmitted via email. The recipient advised to check this email and any attachments for the presence of viruses. Dexterity has taken responsible protection to prevent this risk and accepts no liability for any damage caused by any virus transmitted by this mail.[attachment winmail.dat deleted by Charles P Wright/Watson/IBM] - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 Caution -Disclaimer --- The information contained in the electronic message and any attachments to this message are intended for the exclusive use of the addressee(s)and may contain confidential or privileged information. If you are not intended recipient, please notify the sender immediately and destroy all copies of this message and any attachments. Computer viruses can be transmitted via email. The recipient advised to check this email and any attachments for the presence of viruses. Dexterity has taken responsible protection to prevent this risk and accepts no liability for any damage caused by any virus transmitted by this mail. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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] required clarification
Make sure that you can do nslookup rhc or ping -c 1 rhc. Charles kovendan [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 09/21/2007 03:57 AM To Sipp-users@lists.sourceforge.net cc Subject [Sipp-users] required clarification Hi all, Can anyone help me how to rectify the error Can't get local IP address in getaddrinfo, local_host='rhc', local_ip=''. This the kernel version Linux version 2.6.18-1.2798.fc6 -- regards, kovendan - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 2005. 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] [URGENT] Cannot run sipp in solaris
Also, you may try export TERM=vt100. Charles [EMAIL PROTECTED] wrote on 09/20/2007 05:50:04 AM: You forgot the ?sf option before your scenario name. Make also sure you set your DISPLAY properly. Olivier Boulkroune De : [EMAIL PROTECTED] [mailto:sipp-users- [EMAIL PROTECTED] De la part de Santosh Reddy Envoyé : jeudi 20 septembre 2007 11:20 À : SIPp Cc : [EMAIL PROTECTED]; [EMAIL PROTECTED] Objet : [Sipp-users] [URGENT] Cannot run sipp in solaris Hi all, I am trying to run sipp through SUN SOLARIS OS got error like Error opening terminal: xterm. Please find the below command bash-3.00# ./sipp uas.xml -p 5045 Error opening terminal: xterm. bash-3.00# Can anyone please help me out. Tell me if you need any other information -- Thanks Regards, Santosh Reddy. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 2005. 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] routes last_via fix available in same branch?
Marc, The keyword should be [routes] without the colon. Charles [EMAIL PROTECTED] wrote on 09/13/2007 03:38:31 PM: Hi, I saw a email from Oliver Boulkroune (Re: [Sipp-users] [last_Via:} is dropping characters) on 6-25-07 regarding the fix for last_via dropping characters, so I downloaded the lastest unstable branch sipp.2007-09-13 However, this branch does not have support for the routes: keyword. 2007-09-13 15:27:47:2891189711667.289427: Unsupported keyword 'routes:' in xml scenario file. Does anyone know of a branch (stable or unstable) that contains both? Cheers, Marc- This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 2005. 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] routes last_via fix available in same branch?
Marc, I updated the Wiki to include the proper syntax. Thanks for pointing this out. Charles Marc Archer [EMAIL PROTECTED] wrote on 09/13/2007 03:51:11 PM: thanks Charles! I took the script from http://sipp.sourceforge.net/wiki/index.php/INVITE Should have checked the SIPP documentation to check the correct syntax. Cheers, Marc On 9/13/07, Charles P Wright [EMAIL PROTECTED] wrote: Marc, The keyword should be [routes] without the colon. Charles [EMAIL PROTECTED] wrote on 09/13/2007 03:38:31 PM: Hi, I saw a email from Oliver Boulkroune (Re: [Sipp-users] [last_Via:} is dropping characters) on 6-25-07 regarding the fix for last_via dropping characters, so I downloaded the lastest unstable branch sipp.2007-09-13 However, this branch does not have support for the routes: keyword. 2007-09-13 15:27:47:2891189711667.289427: Unsupported keyword 'routes:' in xml scenario file. Does anyone know of a branch (stable or unstable) that contains both? Cheers, Marc- This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 2005. 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] Large memory usage per TLS connection problem
You could try using a heap profiling tool like massif to see if you can get any ideas. Charles [EMAIL PROTECTED] wrote on 09/11/2007 07:25:52 PM: Hi, Now I am using SIPP TLS connections to test one sip server, but what surprised me is that the memory usage per TLS connection in SIP server side is about 1MB, which is too large. I also tried to use openssl s_client to connect to my SIP server, it only consumed about 30kB per TLS connection. SO i guess the large memory usage is cause by SIPP test tool, am I right? Any one have any idea why I got such unreasonable usage for SIP server with TLS connection? Any configuration that I missed? Or any suggestion about how to solve this problem? Thanks in advance! Carl - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. 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 2005. 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] Example of scenario wher B hangs up
Andreas, It should be relatively straight forward to adapt the standard UAC/UAS scenarios to do this, as long as you want only the UAS to send the bye (if you want to make it 50/50 or something more complicated, you'll need to use 3pcc or some other synchronization mechanism). Just copy and paste the send of the BYE and recv of a 200 from the UAC to the UAS and vice versa. Charles Andreas Byström [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 09/06/2007 07:42 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Example of scenario wher B hangs up Hi all, I have been searching for a XML file that contains a scenario where the B part terminates teh session (sends Bye). Any hints where I can find such a example? Regards, // Andreas - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.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: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] about the expresion
You should try - instead of ~ for the ranges in your regular
expressions (e.g., [0-9a-z]).
Charles
[EMAIL PROTECTED] wrote on 08/30/2007 01:56:52 PM:
hi,
i have a problem, in the scenario,i want to extract this parameter of
the
sip message.
it include this header
Contact:sip:[EMAIL PROTECTED];Dpt=2q5b-13
i want to extract Dpt=2q5b-13
but i can not success to do that
my way:
actionereg regexp=Dpt=[0~9a~z]{4}\-[0~9]{2}* search_in=msg
check_it=true assign_to=1 /
/action
it is fail !! can everyone help me how to write the regexp?
thanks!!
-
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Re: [Sipp-users] How to use sipp to deal with the following case?
This is not easy to accomplish with a standard script, but if you make the decision ahead of time about who will hang up you can probably communicate it from the UAC to the UAS using custom headers or third party call control. Charles [EMAIL PROTECTED] wrote on 08/23/2007 10:48:42 AM: Hi, I want to use sipp to act as a UAC, but both of UAC and UAS can send BYE. What I want is that if UAS sends BYE first, UAC reply 200 OK. if UAC sends BYE before UAS, then wait for the response. if both send BYE almost at the same time, UAC reply 200 OK and wait for the response. In this case, UAC should have higher priority to wait for BYE from UAS. That is, sipp can wait for BYE for a while, but as long as BYE is received, it must send 200 OK immediately. Can sipp deal with this case? If there is a timer for recv command, the problem can be solved somehow. Does anyone have an idea? Thank you in advance. Best, Zheng Da - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.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: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Call variables not expanded
You must use a recent subversion trunk version (at least r275). Also, the more recent versions will complain about bad actions. Charles K L [EMAIL PROTECTED] wrote on 08/10/2007 05:23:07 AM: I'm using the 2.0.1 release. On 8/9/07, Charles P Wright [EMAIL PROTECTED] wrote: The assignstr action was not defined until after the 2.0 release. Charles [EMAIL PROTECTED] wrote on 08/09/2007 12:35:40 PM: I have the same issue when trying to expand (use/apply) stored variable, results comes up blank in the outgoing BYE To field. The assignment is not being made. I'm using version 2.0 official release: nop action assignstr assign_to=3 value=[peer_tag_param] / /action /nop send retrans=500 ![CDATA[ BYE sip:[$2] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port]; branch=ap[branch];received=166.34.101.40 From: sip:[EMAIL PROTECTED]:[local_port];tag=[pid] SIPpTag00[call_number] To: sip:[EMAIL PROTECTED]:[remote_port][$3] Call-ID: [call_id] CSeq: [cseq] BYE User-agent: CS2000/NGSS/7.0 Reason: Q.850; cause=16; text=Normal call clearing Max-Forwards: 69 Contact: sip:[EMAIL PROTECTED]:[local_port] Require: 100rel,replaces Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY, PRACK [routes] Content-Length: 0 ]] K L [EMAIL PROTECTED] wrote: Hello, In my SIPp scenario, I'm trying to At the beginning of the scenario, I have this: ... But both $1 and $2 expand to the empty string when I'm using them (through [$1] and [$2]) in the recv and send blocks. Is that expected ? How to get around this ? Regards, K.L. - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users Pinpoint customers who are looking for what you sell. Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.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: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Call variables not expanded
The assignstr action was not defined until after the 2.0 release. Charles [EMAIL PROTECTED] wrote on 08/09/2007 12:35:40 PM: I have the same issue when trying to expand (use/apply) stored variable, results comes up blank in the outgoing BYE To field. The assignment is not being made. I'm using version 2.0 official release: nop action assignstr assign_to=3 value=[peer_tag_param] / /action /nop send retrans=500 ![CDATA[ BYE sip:[$2] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port]; branch=ap[branch];received=166.34.101.40 From: sip:[EMAIL PROTECTED]:[local_port];tag=[pid] SIPpTag00[call_number] To: sip:[EMAIL PROTECTED]:[remote_port][$3] Call-ID: [call_id] CSeq: [cseq] BYE User-agent: CS2000/NGSS/7.0 Reason: Q.850; cause=16; text=Normal call clearing Max-Forwards: 69 Contact: sip:[EMAIL PROTECTED]:[local_port] Require: 100rel,replaces Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY, PRACK [routes] Content-Length: 0 ]] K L [EMAIL PROTECTED] wrote: Hello, In my SIPp scenario, I'm trying to At the beginning of the scenario, I have this: ... But both $1 and $2 expand to the empty string when I'm using them (through [$1] and [$2]) in the recv and send blocks. Is that expected ? How to get around this ? Regards, K.L. - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users Pinpoint customers who are looking for what you sell. Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.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: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Need more variables
Jaime, Also, what version are you using? The latest trunk versions should support an arbitrary number of variables. Charles Boulkroune, Olivier (Non-HP:Atos Origin) [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 08/06/2007 04:45 AM To Jaime Cabrera [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] Need more variables Hello Jaime, What happens if you set more than 19 variables in your scenario ? Regards, Olivier Boulkroune De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jaime Cabrera Envoyé : lundi 23 juillet 2007 15:58 À : sipp-users@lists.sourceforge.net Objet : [Sipp-users] Need more variables Hello, I need more than 19 variables, ¿how can I have it? Thanks Jaime Cabrera - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.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: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] I'm Beat!!! Need help , Trying to use a variable name as a pcap file to be played by
You can not use a variables for most of the XML parameters, including play_pcap_audio. To make this work you would need to modify SIPp itself as described in the message you cited. Charles [EMAIL PROTECTED] wrote on 07/31/2007 09:35:46 AM: Hi , I've been watching the thread on this issue, and I'm not sure if the ending was that , this kind of variable assignment only works for double, I believe this is whats stated in http://www.mail-archive.com/sipp-users@lists.sourceforge.net/msg01495.html this is the most recent thread I can find related to this issue. The section I'm having the problem with : nop action assignstr assign_to=1 value=[field0] / log message=ONE: [$1] FIELD0:[field0] / exec play_pcap_audio=[$1]/ /action /nop The output is 2007-07-31 15:12:04:0841185891124.084705: Can't open PCAP file '$1]/ /action /nop pause milliseconds='. if I comment out the exec line , the senario works fine, and the logging reveals that the assignment to 1 does happen. I;ve tried every variation of [$1] , $1 , etc. Will there be changes needed to the sipp build to cope with strings in this senario? And does anyone have a work around... Please not, I don't have much xml experience. looking forwardt o getting this working thanks Noel Nesbitt Avaya - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.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: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Desparate Help on assignstr function for variable assignment
Andrew, In the intial action where you are assigning a value to 3, you should verify that [peer_tag_param] itself produces output with a logging action. Charles [EMAIL PROTECTED] wrote on 07/31/2007 10:37:34 AM: Hello members, This is my very first post in this forum. 1 month experience in SIPp. Dealing with a Conference Call Scenario with multiple INVITEs (legs). I need to send a BYE at the end to kill the very first Conf INVITE call leg. To do this I must save the peer_tag_param into a string variable and later, stick it at the end of the To: header of the BYE. I coded the following assignstr function to store the peer_tag_param from the response for Conf INVITE call leg into a variable 3. My problem is that I can't seem to echo this variable 3 nor can I stick the 3 into the BYE. The stringstr assignment is NOT BEING MADE. Or I'm not using var variable correctly. nop action assignstr assign_to=3 value=[peer_tag_param]/ exec command=echo variable=[3]/ç=resulted in: unsupported keywork 3 exec command=echo variable=[$3]/ =resulted in: variable= exec command=echo variable=better work/ çresulted in: variable=better work /action /nop Here's my Sent BYE segment: send retrans=500 ![CDATA[ BYE [EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port]; branch=ap[branch];received=166.34.101.40 From: sip:[EMAIL PROTECTED]:[local_port];tag=[pid] SIPpTag00[call_number] To: sip:[EMAIL PROTECTED]:[remote_port]** WHAT TO PUT HERE [$3]?? Call-ID: [call_id] CSeq: [cseq] BYE User-agent: CS2000/NGSS/7.0 Reason: Q.850; cause=16; text=Normal call clearing Max-Forwards: 69 Contact: sip:[EMAIL PROTECTED]:[local_port] Require: 100rel,replaces Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY, PRACK [routes] Content-Length: 0 ]] /send Seems so simple but have been killing me for over a week. Please help. -Andrew Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.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: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] IM/Presence support at SIPp
I have successfully generated SIMPLE traffic with SIPp, you should be able to easily adapt the default UAC and UAS scenarios by changing the BYE message to a MESSAGE message and removing the INVITE transaction. Charles -- Dr. Charles P. Wright Research Staff Member Network Server Systems Software IBM T.J. Watson Research Center [EMAIL PROTECTED] wrote on 07/26/2007 11:26:55 AM: Hi Yaniv, Could you detail more precisely what kind of call-flow you would like to obtain ? I don?t know much about SIMPLE protocol, but I would say that sipp will support SIMPLE messages as long as they are SIP-compliant. Any feedbacks from your experience would be welcome ! Regards, Olivier Boulkroune De : [EMAIL PROTECTED] [mailto:sipp-users- [EMAIL PROTECTED] De la part de Ben-Hamou Yaniv Envoyé : jeudi 26 juillet 2007 16:25 À : sipp-users@lists.sourceforge.net Objet : [Sipp-users] IM/Presence support at SIPp Hi, I would like to know whether SIPp supports IM/Presence flows (SIP/SIMPLE). It will be very helpful if I could get few client examples. Thanks for supporting. Yaniv. - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.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: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Problem of SIPp and OpenSER in -t tn mode
Zou, If you want a truly open-loop workload generator you should use -l 0, which removes the open call limit. Unfortunately, if the remote host is under overload (i.e. you have exceeded its capacity and calls can not complete at the rate you desire), then SIPp in multi-connection mode is going to inevitably use up the maximum number of sockets available from the OS; which is really a limitation of any workload generator. You may find that you can get better results by running multiple instances of SIPp, as then each one would be able to generate more load before running into limitations. Charles 邹嘉 [EMAIL PROTECTED] wrote on 07/14/2007 06:25:36 AM: Hi, Charles, Thanks for your response. But, in our experiment settings, we set the call limit(-l) to a very large value, to make sure the number of new calls SIPp has made per second equals to the specified call rate. As to the -max_sockets parameter, we've found that after the simultaneous openning sockets number reaches that value, SIPp will not open a new socket for a new call, but reuse a existed socket. In that sense, those conncections becomes persistent, and this is not the situation we want to benchmark. What we want to benchmark is the situation that many clients connect to the OpenSER proxy, and each of those clients opens a connection and will close it after the call finishes. My question is whether SIPp can help us with our benchmark in such situation, thanks very much. Cheers! Zou Jia -Original Message- From: Charles P Wright [EMAIL PROTECTED] Date: Fri, 13 Jul 2007 09:23:54 -0400 To: Boulkroune, Olivier (Non-HP:Atos Origin) [EMAIL PROTECTED] Cc: sipp-users@lists.sourceforge.net, [EMAIL PROTECTED] sourceforge.net, =?GB2312?B?1968zg==?= [EMAIL PROTECTED] Subject: Re: [Sipp-users] Problem of SIPp and OpenSER in -t tn mode Zou, You may also find that setting a call limit (-l) and a maximum number of sockets (-max_multi_socket) will help your situtation. Charles [EMAIL PROTECTED] wrote on 07/13/2007 04:04:45 AM: Hello Zou -t tn implies using TCP with one socket per call, so it's probable you faced sipp/system limitations at high call rate (which call rate ?). If you could avoid using this mode, it will be better. Regards, Olivier Boulkroune ; -Message d'origine- De ;: [EMAIL PROTECTED] [mailto:sipp-users- [EMAIL PROTECTED] De la part de ?? Envoy?;: vendredi 13 juillet 2007 08:17 ?;: sipp-users@lists.sourceforge.net Objet ;: [Sipp-users] Problem of SIPp and OpenSER in -t tn mode Hi, Dear all! ; ;We need to test a scenario that many clients connect to OpenSER proxy server. So, we used the -t tn mode. However, when call rate increases to some level, the SIPp will stop sending packets. Did anybody also encounter such problem? ;Or any suggestions to test such scenario for OpenSER? ; ;Thanks very much! ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ;Yours, ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ; ;Zou Jia - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to code variable length pause range
Don, I have committed some changes that allow you to achieve this functionality. The basic flow is that the [field0] is read into a newly string variable type ($1). $1 is then converted to a double value ($2) [new functionality]. To perform the random sampling, the existing sample action is used with a uniform distribution from zero to one into $3. Finally, $3 is multiplied by $2 [previously you could only add, subtract, multiply, and divide by constants], so you have a double value that is uniformly distributed between zero and [field0]. You can then use this value as the input to a pause command. Sample XML, with some additional logging so that you can track what is going on is below. nop action log message=[call_id]: Started at [clock_tick] / assignstr assign_to=1 value=[field0] / todouble assign_to=2 variable=1 / sample assign_to=3 distribution=uniform min=0 max=1 / log message=[call_id]: Uniform (0, 1): [$3] * [field0] / multiply assign_to=3 variable=2 / /action /nop pause variable=3 / nop action log message=[call_id]: Paused until [clock_tick] / /action /nop Charles -- Dr. Charles P. Wright Research Staff Member Network Server Systems Software IBM T.J. Watson Research Center [EMAIL PROTECTED] wrote on 07/05/2007 12:32:44 PM: I'd like to program a pause that has a variable range. My problem is that I can't figure out how to get the variable into the pause statement. I can set the value of [field0] from a csv file (I verified it by echoing it out). So here is the most obvious way to set the pause range pause distribution=uniform min= max=[field0]/ But I get a Scenario command not implemented error message. So I try to assign it to a numbered variable first: nopactionassign assign_to=1 value=[field0]//action/nop pause distribution=uniform min= max=$1/ This throws the same error. What is the right way to accomplish this? Thanks Don Morrison - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to code variable length pause range
Don, This isn't supported right now, but I'll whip something up that allows you to get it done. Charles [EMAIL PROTECTED] wrote on 07/05/2007 12:32:44 PM: I'd like to program a pause that has a variable range. My problem is that I can't figure out how to get the variable into the pause statement. I can set the value of [field0] from a csv file (I verified it by echoing it out). So here is the most obvious way to set the pause range pause distribution=uniform min= max=[field0]/ But I get a Scenario command not implemented error message. So I try to assign it to a numbered variable first: nopactionassign assign_to=1 value=[field0]//action/nop pause distribution=uniform min= max=$1/ This throws the same error. What is the right way to accomplish this? Thanks Don Morrison - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] How to select different pcap files for different calls
Olivier and Una, The variable manipulation commands only support double values, so you can not assign a string to it. However, based on a few emails in the last week or two, I do think that introducing a string type (which should probably be mostly interchangeable with the regexp type) would be quite useful. This would allow some more flexibility like in the following thread, and also provide the ability to do if statements on strings. Also, for string assignments, using a SendingMessage structure for assignment would be really neat. I didn't see the code where $n is substituted for play_pcap_audio, so I am not sure that the variable will work in this case. For this application though, my suggestion to Una would be to modify the pcap_play_audio attribute to create a SendingMessage structure. That way all of the substitutions could be used. Charles [EMAIL PROTECTED] wrote on 06/22/2007 05:04:11 AM: This is available in the latest unstable versions. Don?t forget to copy to sipp-users mailing-list when replying J Olivier Boulkroune De : RAMSAY,UNA (A-Scotland,ex1) [mailto:[EMAIL PROTECTED] Envoyé : vendredi 22 juin 2007 10:59 À : Boulkroune, Olivier (Non-HP:Atos Origin) Objet : RE: [Sipp-users] How to select different pcap files for different calls HI Olivier Can you please tell me in which version this was added? I am currently using sipp-1.1rc8? Thanks Una From: Boulkroune, Olivier (Non-HP:Atos Origin) [mailto:olivier. [EMAIL PROTECTED] Sent: 22 June 2007 08:39 To: RAMSAY,UNA (A-Scotland,ex1) Cc: sipp-users@lists.sourceforge.net Subject: RE: [Sipp-users] How to select different pcap files for different calls Yes, there is. This variable manipulation feature (see http://sipp. sourceforge.net/doc/reference.html#Actions for more informations) has been recently added. You may try assign assign_to=2 value=[field2) / , although you might probably encounter the same format problem?.. Olivier Boulkroune Tel: +33 4 72 82 37 57 De : RAMSAY,UNA (A-Scotland,ex1) [mailto:[EMAIL PROTECTED] Envoyé : vendredi 22 juin 2007 09:29 À : Boulkroune, Olivier (Non-HP:Atos Origin) Cc : sipp-users@lists.sourceforge.net Objet : RE: [Sipp-users] How to select different pcap files for different calls Hi Olivier Is there a way to assign [field2] to, say, $1? I can see the assign_to but this appears to be part only of the ereg regexp. Thanks Una From: Boulkroune, Olivier (Non-HP:Atos Origin) [mailto:olivier. [EMAIL PROTECTED] Sent: 21 June 2007 14:41 To: RAMSAY,UNA (A-Scotland,ex1) Cc: sipp-users@lists.sourceforge.net Subject: RE: [Sipp-users] How to select different pcap files for different calls You may assign [field2] to a variable $n, and then try something like exec play_pcap_audio=$n/ Olivier Boulkroune De : RAMSAY,UNA (A-Scotland,ex1) [mailto:[EMAIL PROTECTED] Envoyé : jeudi 21 juin 2007 13:16 À : Boulkroune, Olivier (Non-HP:Atos Origin) Cc : sipp-users@lists.sourceforge.net Objet : RE: [Sipp-users] How to select different pcap files for different calls Hi Olivier Thanks for your reply. The external csv file is very powerful and I am already using it to vary the sip From and To fields, where the substitution works fine. However, for the pcap play, the command is expecting a quoted string. I have tried a few scenarios but cannot get sipp to accept this format So far I have tried exec play_pcap_audio=[field2]/ -with field2 in the csv file in the form ;dtmf_2833_5.pcap exec play_pcap_audio=[field2]/ -with field2 in the csv file in the form ;dtmf_2833_5.pcap exec play_pcap_audio=\[field2\]/ -with field2 in the csv file in the form ;dtmf_2833_5.pcap Is there a way of assigning this? Thanks Una From: Boulkroune, Olivier (Non-HP:Atos Origin) [mailto:olivier. [EMAIL PROTECTED] Sent: 21 June 2007 11:34 To: [EMAIL PROTECTED]; sipp-users@lists.sourceforge.net Subject: RE: [Sipp-users] How to select different pcap files for different calls Hi Una, What about using external file fields injection ? Something like nop action exec play_pcap_audio=[field0]/ /action /nop See http://sipp.sourceforge.net/doc/reference. html#Injecting+values+from+an+external+CSV+during+calls for more details. Olivier Boulkroune De : [EMAIL PROTECTED] [mailto:sipp-users- [EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : jeudi 21 juin 2007 12:20 À : sipp-users@lists.sourceforge.net Objet : [Sipp-users] How to select different pcap files for different calls Hello team Can you please tell me if it is possible to select different pcap files for different calls? I would like to do the following call1 - play audio1.pcap call2 - play audio2.pcap etc for approx 30 calls Is there currently any way to do this? I have unique call-ids for each call so I was planning to read that into a variable and then test it to then jump
Re: [Sipp-users] SIP INFO (vfu) after re-INVITE
Marek, I would try inserting an optional receive, and some goto labels. Charles [EMAIL PROTECTED] wrote on 06/20/2007 09:34:46 AM: Hello, I found the option '-aa' which helps me with the INFO issue. However, I have a similar problem with arbitrary number of periodic re-INVITEs (they also arrive during pause) where I would like a similar approach (preferably with a customized response, instead of the built-in one). The '-aa' option covers only INFO, UPDATE and NOTIFY... I think that I could use the '-nd' option but I need to respond to the re-INVITEs instead of ignoring them completely, otherwise the SUT closes the call. Thanks Marek De : [EMAIL PROTECTED] [mailto:sipp-users- [EMAIL PROTECTED] De la part de Shimara, Marek (Altran) Envoyé : mercredi 20 juin 2007 14:13 À : sipp-users@lists.sourceforge.net Objet : [Sipp-users] SIP INFO (vfu) after re-INVITE Hello, I use latest SIPp v2.0-PCAP, version 20070619, built Jun 20 2007, 13:31:07. I have this scenario (based on UAC): Messages Retrans Timeout Unexpected-Msg INVITE -- 1 0 0 100 -- 0 0 0 0 180 -- 0 0 0 0 200 -- E-RTD1 1 0 0 0 ACK -- 1 0 [ NOP ] [ NOP ] INFO -- 1 0 0 200 -- 1 0 Pause [ 4000ms] 1 0 [ NOP ] INVITE -- 1 0 0 200 -- 1 0 ACK -- 1 0 0 INFO -- 1 0 0 200 -- 1 0 Pause [ 1:00] 1 1 BYE -- 0 0 0 200 -- 0 0 0 0 After the INVITE which comes from the SUT I respond with 200 OK and then after the ACK, INFO and 200 OK, I would like to pause (there is PCAP play which was started after the first ACK, still going on during that period). However, after reaching the pause, the call is aborted with 2007-06-20 13:42:31:9221182339751.922596: Aborting call on unexpected message for Call-ID '[EMAIL PROTECTED]': while expecting '0' response, received 'INFO sip: [EMAIL PROTECTED]:5062 SIP/2.0 Via: SIP/2.0/UDP 16.16.93.224:5060; branch=z9hG4bKaba7c1836e027a2fab45fae5ab123fe70106.1 From: sut sip:[EMAIL PROTECTED]:5060; tag=a3e7c024fe279bb02d720bfd3d8f831739dd4fd2 To: sipp_user19540251867482 sip:[EMAIL PROTECTED] 243:5062;tag=1 Call-ID: [EMAIL PROTECTED] CSeq: 4 INFO Content-Type: application/media_control+xml Content-Length:169 Max-Forwards: 70 ?xml version=1.0 encoding=utf-8 ?media_controlvc_primitiveto_encoderpicture_fast_update/picture_fast_update/to_encoder/vc_primitive/media_control I tried to add an optional=true into the recv INFO : ... ... recv request=INFO optional=true action assign assign_to=1 value=1 / /action /recv send test=1 ![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Content-Length: 0 ]] /send pause milliseconds=6/ ... ... But then SIPP never sends the 200 OK response - it stays at Messages Retrans Timeout Unexpected-Msg INVITE -- 1 0 0 100 -- 0 0 0 0 180 -- 0 0 0 0 200 -- E-RTD1 1 0 0 0 ACK -- 1 0 [ NOP ] [ NOP ] INFO -- 1 0 0 200 -- 1 0 Pause [ 4000ms] 1 0 [ NOP ] INVITE -- 1 0 0 200 -- 1 0 ACK -- 1 0 0 INFO -- 130 0 200 -- 0 0 Pause [ 1:00] 0 0 BYE -- 0 0 0 200 -- 0 0 0 0 receiving all the INFO messages, and never reaches the end of the scenario. I would like either to automatically
Re: [Sipp-users] SIP INFO (vfu) after re-INVITE
Marek, I would try something like the following: sipp SUT INVITE -- 100 -- 180 -- 200 -- ACK -- [ NOP ] -- pcap play audio (30 mins) [ NOP ] -- pcap play video (30 mins) Pause [ 4000ms] [ NOP ] -- pcap play DTMF, triggers the first re-INVITE on the SUT INVITE -- Set that invite to optional=global, and see what happens. I don't know if you'll be able to get your pause right, but it is worth a try. I would think that it is certainly possible to do something with variables if you introduce a few new primitives to SIPp itself (e.g., get the current time into a variable). You could then sample how long to pause, generate an absolute time, and then do subtraction between the absolute generated time and now before you actually do the pause. 200 -- ACK -- Pause [30:00] -- here all the INVITEs arrive BYE -- 200 -- Charles Shimara, Marek (Altran) [EMAIL PROTECTED] wrote on 06/20/2007 10:07:18 AM: Hello Charles, Thanks for your reply. Could you please elaborate on your idea? My call flow is: sipp SUT INVITE -- 100 -- 180 -- 200 -- ACK -- [ NOP ] -- pcap play audio (30 mins) [ NOP ] -- pcap play video (30 mins) Pause [ 4000ms] [ NOP ] -- pcap play DTMF, triggers the first re-INVITE on the SUT INVITE -- 200 -- ACK -- Pause [30:00] -- here all the INVITEs arrive BYE -- 200 -- The problem is, during the second Pause (30 min) there is a re- INVITE every 2min30. I could just wait for an INVITE 12 times and deal with each of them, but 2min30 is only the default case, in other words it can be set to anything at all... Thanks Marek De : Charles P Wright [mailto:[EMAIL PROTECTED] Envoyé : mercredi 20 juin 2007 15:42 À : Shimara, Marek (Altran) Cc : sipp-users@lists.sourceforge.net; [EMAIL PROTECTED] sourceforge.net Objet : Re: [Sipp-users] SIP INFO (vfu) after re-INVITE Marek, I would try inserting an optional receive, and some goto labels. Charles [EMAIL PROTECTED] wrote on 06/20/2007 09:34:46 AM: Hello, I found the option '-aa' which helps me with the INFO issue. However, I have a similar problem with arbitrary number of periodic re-INVITEs (they also arrive during pause) where I would like a similar approach (preferably with a customized response, instead of the built-in one). The '-aa' option covers only INFO, UPDATE and NOTIFY... I think that I could use the '-nd' option but I need to respond to the re-INVITEs instead of ignoring them completely, otherwise the SUT closes the call. Thanks Marek De : [EMAIL PROTECTED] [mailto:sipp-users- [EMAIL PROTECTED] De la part de Shimara, Marek (Altran) Envoyé : mercredi 20 juin 2007 14:13 À : sipp-users@lists.sourceforge.net Objet : [Sipp-users] SIP INFO (vfu) after re-INVITE Hello, I use latest SIPp v2.0-PCAP, version 20070619, built Jun 20 2007,13:31:07. I have this scenario (based on UAC): Messages Retrans Timeout Unexpected-Msg INVITE -- 1 0 0 100 -- 0 0 0 0 180 -- 0 0 0 0 200 -- E-RTD1 1 0 0 0 ACK -- 1 0 [ NOP ] [ NOP ] INFO -- 1 0 0 200 -- 1 0 Pause [ 4000ms] 1 0 [ NOP ] INVITE -- 1 0 0 200 -- 1 0 ACK -- 1 0 0 INFO -- 1 0 0 200 -- 1 0 Pause [ 1:00] 1 1 BYE -- 0 0 0 200 -- 0 0 0 0 After the INVITE which comes from the SUT I respond with 200 OK and then after the ACK, INFO and 200 OK, I would like to pause (there is PCAP play which was started after
Re: [Sipp-users] [Need Help] Question of variable manipulation
Leo, The variable manipulation support is only available in trunk. Charles [EMAIL PROTECTED] wrote on 06/15/2007 02:55:59 AM: Dear Charles, Thanks for your response. I tried your vartest.xml, but the variable assignment still not working. debian:~/sipp-2.0.1.src# ./sipp -m 1 -sf vartest.xml localhost -trace_logs [ produces output saying that it didn't expect to receive the message it sent itself ] debian:~/sipp-2.0.1.src# cat *.log $5: I am using SIPp 2.0.1 version installed on debian linux system. I also test the xml file on windows xp using SIPp 2.0.1, and get the same result. Is there anything I missed ? Leo Hu MediaTek Inc. WCP / Software Engineering Div.1 Taipei, Taiwan 胡晉華 Leo Hu E-mail: [EMAIL PROTECTED] Tel: +886-2-26598088 ext 6202 Charles P Wright [EMAIL PROTECTED] Charles P Wright [EMAIL PROTECTED] 2007/06/15 下午 12:14 To [EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net, [EMAIL PROTECTED] Subject Re: [Sipp-users] [Need Help] Question of variable manipulation Leo, I am glad to see that this feature would be useful for you. 1. I add $4 by 3, but in log it is still 0. The code doesn't handle adding values to regular expressions (strings). There should be some better error handling, or possibly even automatic type casting added. 2. I assign $5 to 1, and add by 2, but in log, the variable seems no value at all. This is actually puzzling. I have attached a simple scenario that I ran to test this: $./sipp -m 1 -sf vartest.xml localhost -trace_logs [ produces output saying that it didn't expect to receive the message it sent itself] $ cat *.log $5: 3.00 3. I assign the test result to $6, but the variable seems no value at all. There is no code for printing the result of a Boolean variable. 4. BTW, from the document, the variables are floating point values, but in what data type the result from regular expression is stored since the result may be a string. This is why the addition for $4 isn't working correctly. I will try work up a patch that addresses these shortcomings. Charles * Email Confidentiality Notice The information contained in this e-mail message (including any attachments) may be confidential, proprietary, privileged, or otherwise exempt from disclosure under applicable laws. It is intended to be conveyed only to the designated recipient(s). Any use, dissemination, distribution, printing, retaining or copying of this e-mail (including its attachments) by unintended recipient(s) is strictly prohibited and may be unlawful. If you are not an intended recipient of this e-mail, or believe that you have received this e-mail in error, please notify the sender immediately (by replying to this e-mail), delete any and all copies of this e-mail (including any attachments) from your system, and do not disclose the content of this e-mail to any other person. Thank you! [attachment vartest.xml deleted by Charles P Wright/Watson/IBM] - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] [Need Help] Question of variable manipulation
Leo, I am glad to see that this feature would be useful for you. 1. I add $4 by 3, but in log it is still 0. The code doesn't handle adding values to regular expressions (strings). There should be some better error handling, or possibly even automatic type casting added. 2. I assign $5 to 1, and add by 2, but in log, the variable seems no value at all. This is actually puzzling. I have attached a simple scenario that I ran to test this: $./sipp -m 1 -sf vartest.xml localhost -trace_logs [ produces output saying that it didn't expect to receive the message it sent itself] $ cat *.log $5: 3.00 3. I assign the test result to $6, but the variable seems no value at all. There is no code for printing the result of a Boolean variable. 4. BTW, from the document, the variables are floating point values, but in what data type the result from regular expression is stored since the result may be a string. This is why the addition for $4 isn't working correctly. I will try work up a patch that addresses these shortcomings. Charles vartest.xml Description: Binary data - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Question on random branching
Andreas, Do you have a label at the end of your scenario that you jump to after your standard call setup? Something like: recv request=INVITE crlf=true next=1 chance=0.01 /recv send 100 / send 180 / send 200 next=2 / label id=1 / send 500 / label id=2 / Assuming this isn't it if you post your full scenario, then you should be able to get better answers. Charles [EMAIL PROTECTED] wrote on 06/08/2007 04:47:32 AM: Hi all, I have been browsing through the mail archive but cant find any answer to the problem I have. I hope that someone can help me I'm writing a AUS scenario and it starts with receiving an Invite. Then I want that in 99% of the cases the call should be setup with 180, 200 and so on. However, for 1% of the calls I want to send 500 Server Error instead (to simulate a specific case). When reading the manual I found the chance command and tried to using that. This is how my script starts: recv request=INVITE crlf=true next=1 chance=0.01 /recv Now label=1 is jumping to a place where I send 500, and if I dont jump the script continues with setting up a simple call. The problem is when I run this scenario Iget the 500 in all cases, not only in 1% of the cases. Looking again in the documentaiton I can only find examples where test and chance is used togheter. But it also says that test and chance can be combined, i.e to have... so I guess I should be able to use chance alone? If this is not possible, can I add a test that is always true? Thanks in advance! // Andreas - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Escape character
Try \x5B The \x followed by two hex digits is translated into a literal byte value, 5B corresponding to '['. Charles [EMAIL PROTECTED] wrote on 06/08/2007 05:09:58 AM: folks, I want to put the following in to a SIPP script User-Agent: IP Phone [0.1.70] SIPp obviously does not like the [] round the text as it tries to treat it as a variable. Tried using \ and \\ as the escape character but no joy - what can I use to ensure SIPp treats [0.1.70] as a string only. Steve. - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Conditional branching
Eugen, I am curious as to what your use case for a regular expression from the input file would be? That is, I am not sure exactly what you would like SIPp to do. Charles Eugen [EMAIL PROTECTED] wrote on 05/31/2007 10:44:06 AM: Thanks Peter and Charles. I?ve tried ?test? and ?next? with ?pause?, it works fine and it solves my problem. Yes indeed, these attributes should be added to ?nop? as well. Use a configuration file as input for parsing would also be nice to have. Apart from these little problems I?ve found after reading the doc and trying some tests, this is a nice and very useful tool. Thanks to all contributors J Eugen -Original Message- From: Charles P Wright [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 30, 2007 2:57 PM To: Peter Higginson Cc: 'Eugen'; sipp-users@lists.sourceforge.net; sipp-users- [EMAIL PROTECTED] Subject: Re: [Sipp-users] Conditional branching Regarding #1, you should be able to even put your test on a nop, if you can't then that is probably a feature/misdesign that needs to be added/fixed. Charles [EMAIL PROTECTED] wrote on 05/30/2007 01:36:02 PM: 1) Not directly, however there is nothing to stop you putting a test on a short pause (although I have never done it) which would give you a two way split every time. I have put a test both on reception and on a 200OK answer. 2) No, if you want to test a previous variable just use another name. If what you are wanting is the action to be done or not depending on the test, then you really want to add another option as part of the action logic. 3) If you setup the regexp to give a yes/no answer you can do this ? one of the examples matches a To: field in this way. 4) I don?t know (which means probably not). Peter From: [EMAIL PROTECTED] [mailto:sipp-users- [EMAIL PROTECTED] On Behalf Of Eugen Sent: 30 May 2007 15:14 To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] Conditional branching Hi, I?m trying to build a sip registrar stub with sipp. Related to this I have couple of questions that are related somehow: 1. Is it possible with this tool to jump to different labels depending on the variable that was set? Something like a switch rather than an if. 2. Is it possible to do the test (test=?n?) before the action is executed? 3. Is it possible to respond to a message depending on its content? E.g. respond to a REGISTER with 2xx, 4xx, ? depending on the user portion of To: header. 4. Is it possible to perform a regexp with the content of ?inf file? Thanks, Eugen This communication is confidential and is intended solely for the addressee(s). The information contained herein should be considered Confidential Information for purposes of any Non-Disclosure or similar Agreement between Blueslice Networks and the recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without reading, printing, copying or disclosing it. La présente communication est confidentielle et strictement réservée au(x) destinataire(s). L?information ci-jointe doit être considérée comme Information Confidentielle aux fins de tout accord de non- divulgation ou autre entente entre Blueslice Networks et le destinataire. Tous visionnements, utilisations, divulgations ou distributions non autorisés sont interdits. Dans l?éventualité où ce message vous a été envoyé par erreur, veuillez s?il-vous-plait notifier l?émetteur par réponse à ce courriel et veuillez détruire ce message sans le lire, l?imprimer, le copier ou en divulguer le contenu. - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and delete this e-mail. Any unauthorized copying, disclosure or distribution of the contents in this e-mail is strictly forbidden. - Newport Networks Limited is registered in England. Registration number 4067591. Registered office: 6 St. Andrew Street, London EC4A 3LX - - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users
Re: [Sipp-users] Conditional branching
Regarding #1, you should be able to even put your test on a nop, if you can't then that is probably a feature/misdesign that needs to be added/fixed. Charles [EMAIL PROTECTED] wrote on 05/30/2007 01:36:02 PM: 1) Not directly, however there is nothing to stop you putting a test on a short pause (although I have never done it) which would give you a two way split every time. I have put a test both on reception and on a 200OK answer. 2) No, if you want to test a previous variable just use another name. If what you are wanting is the action to be done or not depending on the test, then you really want to add another option as part of the action logic. 3) If you setup the regexp to give a yes/no answer you can do this ? one of the examples matches a To: field in this way. 4) I don?t know (which means probably not). Peter From: [EMAIL PROTECTED] [mailto:sipp-users- [EMAIL PROTECTED] On Behalf Of Eugen Sent: 30 May 2007 15:14 To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] Conditional branching Hi, I?m trying to build a sip registrar stub with sipp. Related to this I have couple of questions that are related somehow: 1. Is it possible with this tool to jump to different labels depending on the variable that was set? Something like a switch rather than an if. 2. Is it possible to do the test (test=?n?) before the action is executed? 3. Is it possible to respond to a message depending on its content? E.g. respond to a REGISTER with 2xx, 4xx, ? depending on the user portion of To: header. 4. Is it possible to perform a regexp with the content of ?inf file? Thanks, Eugen This communication is confidential and is intended solely for the addressee(s). The information contained herein should be considered Confidential Information for purposes of any Non-Disclosure or similar Agreement between Blueslice Networks and the recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without reading, printing, copying or disclosing it. La présente communication est confidentielle et strictement réservée au(x) destinataire(s). L?information ci-jointe doit être considérée comme Information Confidentielle aux fins de tout accord de non- divulgation ou autre entente entre Blueslice Networks et le destinataire. Tous visionnements, utilisations, divulgations ou distributions non autorisés sont interdits. Dans l?éventualité où ce message vous a été envoyé par erreur, veuillez s?il-vous-plait notifier l?émetteur par réponse à ce courriel et veuillez détruire ce message sans le lire, l?imprimer, le copier ou en divulguer le contenu. - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and delete this e-mail. Any unauthorized copying, disclosure or distribution of the contents in this e-mail is strictly forbidden. - Newport Networks Limited is registered in England. Registration number 4067591. Registered office: 6 St. Andrew Street, London EC4A 3LX - - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Handling 2 INVITEs with different Call-ID
Ashok, You can use two different call-ids, but they need to be of the form prefix1///[call_id] and prefix2///[call_id]. Charles [EMAIL PROTECTED] wrote on 05/03/2007 08:57:28 AM: Hi all, Can we send 2 INVITE with different Call-ID from the same sipp script(Basically can one sipp instance handle multiple Call-IDs)?? Plz help on this. - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] problems compiling rev 205 and later
Enrico, No unit testing, but interestingly my STL headers managed to pull in assert.h without me doing it so it compiled on my RHEL4 derived distribution. New fix checked in. Thanks for trying this out and having the fortitude to put up with these errors, Charles Enrico Hartung [EMAIL PROTECTED] wrote on 05/03/2007 11:09:34 AM: Hi, the error is solved, but here's another one: actions.cpp: In member function `void CAction::setAction(CAction)': actions.cpp:303: error: `assert' undeclared (first use this function) actions.cpp:303: error: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [actions.o] Error 1 looks like you're using a unit test framework ... --Enrico Charles P Wright wrote: Enrico, I apologize for the compile error. I am not yet used to SVN and forgot to add these two files to the repository before I ran SVN commit. You should be able to update and compile now. Charles [EMAIL PROTECTED] wrote on 05/03/2007 09:39:48 AM: Hi, the last revision in the repository I can compile with 'make ossl' is 204. All later revisions run into errors: here is the latest error message(r214): In file included from scenario.cpp:30: sipp.hpp:64:22: infile.hpp: No such file or directory In file included from scenario.cpp:30: sipp.hpp:263: error: `FileContents' was not declared in this scope sipp.hpp:263: error: parse error before `' token sipp.hpp:264: error: syntax error before `;' token cheers, Enrico [attachment smime.p7s deleted by Charles P Wright/Watson/IBM] - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] problems compiling rev 205 and later
Enrico,
This actually has me very confused, the file is opened exactly the same as
before:
FileContents::FileContents(const char *fileName) {
ifstream *inFile= new ifstream(fileName);
There is no code for file patterns or anything like that, so I am
wondering if maybe the shell is doing something that is confusing it in
ways that it wasn't confused before. Do you have a script for running
SIPp that makes use of wildcards or something similar?
Charles
Enrico Hartung [EMAIL PROTECTED] wrote on 05/03/2007 11:52:48 AM:
Hi Charles,
it's me again ;-)
Now I can compile sipp without errors.
But I have some problems with your new feature (multiple infiles):
I'm using sipp in a test framework. This framework is generating the
include files for sipp before starting sipp. Therefor it needs template
files like include.csv.tmpl which is the template for include.csv.
Now when I wanna include include.csv into sipp it loads
include.csv.tmpl instead. So I guess sipp is no more checking the
complete file name (exact match), right?
I hope this behavior is only a bug and not needed by your feature ... is
it possible to change it back to exact match checking?
--Enrico
Charles P Wright wrote:
Enrico,
No unit testing, but interestingly my STL headers managed to pull in
assert.h without me doing it so it compiled on my RHEL4 derived
distribution. New fix checked in.
Thanks for trying this out and having the fortitude to put up with
these errors,
Charles
Enrico Hartung [EMAIL PROTECTED] wrote on 05/03/2007 11:09:34 AM:
Hi,
the error is solved, but here's another one:
actions.cpp: In member function `void CAction::setAction(CAction)':
actions.cpp:303: error: `assert' undeclared (first use this
function)
actions.cpp:303: error: (Each undeclared identifier is reported only
once for
each function it appears in.)
make[1]: *** [actions.o] Error 1
looks like you're using a unit test framework ...
--Enrico
Charles P Wright wrote:
Enrico,
I apologize for the compile error. I am not yet used to SVN and
forgot to add these two files to the repository before I ran SVN
commit. You should be able to update and compile now.
Charles
[EMAIL PROTECTED] wrote on 05/03/2007
09:39:48 AM:
Hi,
the last revision in the repository I can compile with 'make
ossl' is
204. All later revisions run into errors:
here is the latest error message(r214):
In file included from scenario.cpp:30:
sipp.hpp:64:22: infile.hpp: No such file or directory
In file included from scenario.cpp:30:
sipp.hpp:263: error: `FileContents' was not declared in this
scope
sipp.hpp:263: error: parse error before `' token
sipp.hpp:264: error: syntax error before `;' token
cheers,
Enrico
[attachment smime.p7s deleted by Charles P Wright/Watson/IBM]
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Re: [Sipp-users] problems compiling rev 205 and later
Fixed. Simple off by one error.
Charles
Enrico Hartung [EMAIL PROTECTED] wrote on 05/03/2007 12:23:29 PM:
Hi,
I was wrong. Sipp loads the correct file. But the behavior is still
different than I expect:
The template file:
SEQUENTIAL
[#U1_USERNAME#];[#U1_DOMAIN#];[authentication username=[#U1_AUTHNAME#]
password=[#U1_PASSWORD#]];
[#U2_USERNAME#];[#U2_DOMAIN#];[authentication username=[#U2_AUTHNAME#]
password=[#U2_PASSWORD#]];
The generated file (I only need one user in this scenario):
SEQUENTIAL
testuser001;osser.sip-router.org;[authentication username=testuser001
password=xxx];
[#U2_USERNAME#];[#U2_DOMAIN#];[authentication username=[#U2_AUTHNAME#]
password=[#U2_PASSWORD#]];
sipp r204 takes the first line (as I expect), but sipp r216 takes the
second line ...
(I thought that sipp loads the template file because I saw the wildcards
of the second line in the sip trace)
Do you know why sipp is using the second line instead of the first one,
although 'sequential' is used?
I hope you understand my problem ...
--Enrico
Charles P Wright wrote:
Enrico,
This actually has me very confused, the file is opened exactly the
same as before:
FileContents::FileContents(const char *fileName) {
ifstream *inFile= new ifstream(fileName);
There is no code for file patterns or anything like that, so I am
wondering if maybe the shell is doing something that is confusing it
in ways that it wasn't confused before. Do you have a script for
running SIPp that makes use of wildcards or something similar?
Charles
Enrico Hartung [EMAIL PROTECTED] wrote on 05/03/2007 11:52:48 AM:
Hi Charles,
it's me again ;-)
Now I can compile sipp without errors.
But I have some problems with your new feature (multiple infiles):
I'm using sipp in a test framework. This framework is generating the
include files for sipp before starting sipp. Therefor it needs
template
files like include.csv.tmpl which is the template for
include.csv.
Now when I wanna include include.csv into sipp it loads
include.csv.tmpl instead. So I guess sipp is no more checking the
complete file name (exact match), right?
I hope this behavior is only a bug and not needed by your feature
... is
it possible to change it back to exact match checking?
--Enrico
Charles P Wright wrote:
Enrico,
No unit testing, but interestingly my STL headers managed to pull
in
assert.h without me doing it so it compiled on my RHEL4 derived
distribution. New fix checked in.
Thanks for trying this out and having the fortitude to put up with
these errors,
Charles
Enrico Hartung [EMAIL PROTECTED] wrote on 05/03/2007 11:09:34 AM:
Hi,
the error is solved, but here's another one:
actions.cpp: In member function `void
CAction::setAction(CAction)':
actions.cpp:303: error: `assert' undeclared (first use this
function)
actions.cpp:303: error: (Each undeclared identifier is reported
only
once for
each function it appears in.)
make[1]: *** [actions.o] Error 1
looks like you're using a unit test framework ...
--Enrico
Charles P Wright wrote:
Enrico,
I apologize for the compile error. I am not yet used to SVN
and
forgot to add these two files to the repository before I ran
SVN
commit. You should be able to update and compile now.
Charles
[EMAIL PROTECTED] wrote on 05/03/2007
09:39:48 AM:
Hi,
the last revision in the repository I can compile with 'make
ossl' is
204. All later revisions run into errors:
here is the latest error message(r214):
In file included from scenario.cpp:30:
sipp.hpp:64:22: infile.hpp: No such file or directory
In file included from scenario.cpp:30:
sipp.hpp:263: error: `FileContents' was not declared in this
scope
sipp.hpp:263: error: parse error before `' token
sipp.hpp:264: error: syntax error before `;' token
cheers,
Enrico
[attachment smime.p7s deleted by Charles P
Wright/Watson/IBM]
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Re: [Sipp-users] SIPp 2.0 truncating last_via by 1 character
Tarek, I believe it was the short form header patch that I posted that broke this. Another user had the same issue and I sent them this patch, but never got any feedback. Does this fix the issue for you? Charles [EMAIL PROTECTED] wrote on 05/01/2007 03:38:56 PM: Hi, I've recently migrated to SIPp 2.0 (official) and have observed the following issue. The 200 response, which is built using [last_via], when the Via contains multiple headers, is bring truncated by a single character when re-built. Here's the trace - notice the first branch= tag is missing the final digit. Olivier, you sent a patch out just a day or two before 2.0 was pulled regarding last_via... perhaps some breakage ? (I was previously using the Dec 08 build, so its possible this has been broken a while, but the last_via patch seems a good candidate) t NOTIFY sip:[EMAIL PROTECTED]:5060;transport=tcp SIP/2.0^M Via: SIP/2.0/TCP 67.1.100.81:5060; branch=z9hG4bK7c957453-37910fda-4fe40904-5a984882-1^M Record-Route: sip:user-01-01. [EMAIL PROTECTED]:5060;maddr=67.1.100.81;lr^M From: sip:[EMAIL PROTECTED];tag=da5472e4^M To: sip:[EMAIL PROTECTED];tag=193961-3^M CSeq: 1073741824 NOTIFY^M Call-ID: 3-///[EMAIL PROTECTED] Event: presence^M User-Agent: Cisco-PE/6.0.1.1^M Contact: sip:67.1.100.81:5070;transport=tcp^M Content-Length: 7624^M Content-Type: multipart/related;type=application/rlmi+xml; start=[EMAIL PROTECTED];boundary=da56a10e-1dd1-11b2-b^M Require: eventlist^M Subscription-State: active;expires=86400^M Via: SIP/2.0/TCP 67.1.100.81:5070;received=67.1.100.81; branch=z9hG4bKda5740be-1dd1-11b2-b908-83397ec92f4f^M Max-Forwards: 68^M SIP/2.0 200 OK^M Via: SIP/2.0/TCP 67.1.100.81:5060; branch=z9hG4bK7c957453-37910fda-4fe40904-5a984882-, SIP/2.0/TCP 67. 1.100.81:5070;received=67.1.100.81;branch=z9hG4bKda5740be-1dd1-11b2- b908-83397ec92f4f^M From: sip:[EMAIL PROTECTED];tag=da5472e4^M To: sip:[EMAIL PROTECTED];tag=193961-3^M Call-ID: 3-///[EMAIL PROTECTED] CSeq: 1073741824 NOTIFY^M Contact: sip:67.1.100.70:5060;transport=TCP^M Content-Length: 0^M - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users headerctrlm.diff Description: Binary data - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Problem with recv
Alice's message will have a different Call-ID than Bob's existing call. Therefore the message will not be identified as part of the conversation. You can use three or fou r SIPp instances for this to work. 1. Bob's Registration 2. Bob's Message Receive 3. Alice (or Alice's Registration and Send) Charles [EMAIL PROTECTED] wrote on 04/26/2007 01:50:26 AM: Hello I have two sipp scrips representing alice and bob. When I start the alice script, it registers to the SIP server, sends a SIP:MESSAGE to bob trough the same sip server and then, when 200 OK is received, it unregisters from the server. Bob will also register to the same SIP server, then it waits for the message from alice, sends the 200 OK and then unregisters from the server. Here is the flow: Alice SIP server Bob ||| | REGISTER | REGISTER | |---|---| |401 | 401| |---|---| ||| ||| | REGISTER | REGISTER | |---|---| |401 | 401| |---|---| ||| ||| | MESSAGE | MESSAGE | |---|---| |200 | 200| |---|---| ||| ||| ||| | REGISTER | REGISTER | |---|---| |200 | 200| |---|---| The problem is that Bob's sipp scrip does not parse the received SIP:MESSAGE as it should. I can see the message in the messages log, but the script does not want to jump over the recv block. What can the problem be? Thank you LucTeo Here are the scripts: alice.xml - ?xml version=1.0 encoding=ISO-8859-1 ? !DOCTYPE scenario SYSTEM sipp.dtd !-- Register -- scenario name=alice_sends_message_to_bob send retrans=500 ![CDATA[ REGISTER sip:[field2] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] sip:[EMAIL PROTECTED] ;tag=[call_number] To: [field0] sip:[EMAIL PROTECTED] P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E Call-ID: reg///[call_id] CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:[local_port] Max-Forwards: 20 Expires: 1800 User-Agent: Sipp v1.1-TLS, version 20061124 Content-Length: 0 Supported: path ]] /send recv response=401 auth=true rtd=true action ereg regexp=.* search_in =hdr header=Service-Route assign_to= 1 / /action /recv send retrans=500 ![CDATA[ REGISTER sip:[field2] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Route: [$1] From: [field0] sip:[EMAIL PROTECTED];tag=[call_number] To: [field0] sip:[EMAIL PROTECTED] P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E Call-ID: [call_id] CSeq: 2 REGISTER Contact: sip:[EMAIL PROTECTED]:[local_port] [field3] Max-Forwards: 20 Expires: 1800 User-Agent: Sipp v1.1-TLS, version 20061124 Content-Length: 0 Supported: path ]] /send recv response=100 optional=true / recv recv response=200 / recv !-- Send message and wait for response -- pause milliseconds =500 / send retrans=500 ![CDATA[ MESSAGE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: [field0] sip:[EMAIL PROTECTED] ;tag=[call_number] To: [field1] sip:[EMAIL PROTECTED] P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E Call-ID: msg-[call_id] CSeq: 1 MESSAGE Max-Forwards: 20 User-Agent: Sipp v1.1-TLS, version 20061124 Content-Type: text/plain Content-Length: [len] Hello Bob! ]] /send recv response=100 optional=true /recv recv response=200 /recv !-- Unregister -- send retrans=500 ![CDATA[ REGISTER sip:[field2] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Route: [$1] From: [field0] sip:[EMAIL PROTECTED];tag=[call_number] To: [field0] sip:[EMAIL PROTECTED] P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E Call-ID: [call_id] CSeq: 3 REGISTER Contact: sip:[EMAIL PROTECTED]:[local_port] [field3] Max-Forwards: 20 Expires: 1800 User-Agent: Sipp v1.1-TLS, version 20061124 Content-Length: 0 Supported: path ]] /send recv response=100 optional=true /recv recv response=200 action log message=SUCCESS/ /action /recv ResponseTimeRepartition
Re: [Sipp-users] Receivinf REGISTER and INVITE in the same sipp script
Ashok, The REGISTER and INVITE will have different Call-IDs. To make this work you need to use optional messages, so that either a register or an invite can start a new scenario. Charles [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 04/26/2007 08:31 AM To sipp-users@lists.sourceforge.net cc [EMAIL PROTECTED], [EMAIL PROTECTED] Subject [Sipp-users] Receivinf REGISTER and INVITE in the same sipp script Hi, I am running a sipp script in which sipp is receiving REGISTER, sending 200 OK, and receiving INVITE. When I am trying to receive INVITE then sipp reports it as an unexpected message. Can you please help on this. The folowing Error occurs: 2007-04-26 18:05:15: Aborting call on unexpected message for Call-ID '[EMAIL PROTECTED]': while expecting 'REGISTER', received 'INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0 Here is the sipp script: ?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 'uas' scenario. -- !-- -- scenario name=branch_server !-- By adding rrs=true (Record Route Sets), the route sets -- !-- are saved and used for following messages sent. Useful to test -- !-- against stateful SIP proxies/B2BUAs. -- recv request=REGISTER /recv send ![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Content-Length: 0 ]] /send recv request=INVITE crlf=true action ereg regexp=sut search_in=hdr header=From: assign_to=3/ /action /recv !-- The '[last_*]' keyword is replaced automatically by the -- !-- specified header if it was present in the last message received -- !-- (except if it was a retransmission). If the header was not -- !-- present or if no message has been received, the '[last_*]' -- !-- keyword is discarded, and all bytes until the end of the line -- !-- are also discarded. -- !-- -- !-- If the specified header was present several times in the -- !-- message, all occurences are concatenated (CRLF seperated) -- !-- to be used in place of the '[last_*]' keyword. -- send next=1 test=3 ![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:] [last_Call-ID:] [last_CSeq:] Content-Length: 0 ]] /send !-- Keep the call open for a while in case the 200 is lost to be -- !-- able to retransmit it if we receive the BYE again. -- pause milliseconds=4000/ !-- definition of the response time repartition table (unit is ms) -- ResponseTimeRepartition value=10, 20, 30, 40, 50, 100, 150, 200/ !-- definition of the call length repartition table (unit is ms) -- CallLengthRepartition value=10, 50, 100, 500, 1000, 5000, 1/ /scenario And the corresponding log file is: -- Scenario Screen [1-9]: Change Screen -- Port Total-time Total-calls Transport 5061 15.32 s2 UDP 0 new calls during 0.320 s period 4 ms scheduler resolution 1 callsPeak was 2 calls, after 2 s 0 Running, 1 Paused, 0 Woken up 1 open sockets Messages Retrans Timeout Unexpected-Msg -- REGISTER 1 0 1 -- 200 1 0 -- INVITE0 00 -- 200 0 0 [ 4000ms] Pause 0 0 - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2
Re: [Sipp-users] An extra comma in 2nd via when use [last_Via] handle short form
Joseph, Can you try the attached patch? Charles [EMAIL PROTECTED] wrote on 04/26/2007 03:57:07 PM: I have loaded the latest sipp version and found an extra comma in the 2nd via header when use [Last_Via] handle the short form message. pl check the 180 Ringing message below. Thanks, Joseph 2007-04-17 15:11 oboulkroune * call.cpp, call.hpp: Fix: updated support of short header forms - provided by Charles P. Wright from IBM Research Protocol : SIP-2.0 SIP/2.0 180 RingingCRLF v: SIP/2.0/UDP 166.35.250.121:5060;branch=z9hG4bK7d124e16f513aa6848f1d1f1076e1937.a864a1fCR ,v: SIP/2.0/UDP 166.35.139.68:5041;branch=z9hG4bK-18687-1-0;received=166.35.139.68CRLF f: sip:[EMAIL PROTECTED]:5041;tag=18687SIPpTag001CRLF t: sip:[EMAIL PROTECTED]:5060;tag=18684SIPpTag011CRLF i: [EMAIL PROTECTED]CRLF CSeq: 1 INVITECRLF Record-Route: sip:166.35.250.121;lrCRLF Contact: sip:166.35.139.68:50457;transport=UDPCRLF Content-Length: 0CRLF CRLF __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users headerctrlm.diff Description: Binary data - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Changes for using SIPp in an implementation of ETSI TISPAN IMS Benchmark
Peter Higginson [EMAIL PROTECTED] wrote on 04/25/2007 03:35:06 AM: One important thing to consider here is that we did not take care so far of other protocols/transports than UDP (we think our design will allow an easy integration of the others, but we did not 'port' the TCP and TLS implementations over just yet). So a combination of our efforts will hopefully be very beneficial. OK. I look forward to seeing your patch. One of the most important things that we did was to get all of the data for a single socket in a structure, and create choke point functions for sending and receiving, which is especially important for TCP and TLS (which both need to deal with congestion, partial messages, and multiple messages). Multiple IP addresses or ports with TCP is quite complex and you definitely need a structure/class to represent each open TCP call. Correct SIP protocol requires that responses use the same call as the corresponding request (if still open) but for requests you have to find or open a call to the address/port given in the last contact. (Simple calls are fine ? it is things like OPTIONS and re- INVITE by the called party that get messy.) That means remembering which TCP call a request arrived on and on sending requests scanning all the TCP calls in use to find if there is one already open that you can use or opening one. One relatively large data structure improvement in there is that there is a correct mapping of calls to sockets and vice versa. I did this by always setting the call-call_socket to the actually used socket, throwing in some reference counting and a multimap from sockets to calls. The network code that I have does not attempt to do the socket matching, but it would not be very difficult to extend it to do so using a map that maps a (protocol, ip, port) tuple to a struct sipp_socket *. I also found TCP open taking up to 30 seconds (usually due to socket reuse timeouts) so you cannot just freeze SIPP till a call opens, you have to keep state and poll. I did not handle the blocking connect, or the blocking close case (the latter becomes important if the socket is congested). If you have code for non-blocking connect or close, I think that would be an important feature to add. The multiple IP address code we have for UDP was taken from another project so I was never able to share it. (It is very messy anyway so you probably would not want it.) However I had to re-write the socket management to get TCP to work so there is no problem sharing that part and the changes to the SIPP files if that would help. (It does not include TLS though.) I don't think that it could hurt. :) -- Dr. Charles P. Wright Research Staff Member Network Server Systems Software IBM T.J. Watson Research Center- This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Changes for using SIPp in an implementation of ETSI TISPAN IMS Benchmark
Dragos Vingarzan [EMAIL PROTECTED] wrote on 04/25/2007 03:00:46 AM: Hello Charles, David, Olivier, We at FOKUS, for a 3rd party :), just started about a month ago too, working on such tools for benchmarking. However, we took a different approach as we believe that there is a true value in having even a light-weight SIP stack in SIPp (regexp are great, but if you need to get a lot of info from messages, they might be less efficient, easy-to-use and safe than parsing one time). There are others in my group (Erich in particular) who also agree that a SIP stack might be the way to go. There is certainly more need for SIP knowledge (e.g., the retransmission hash needs to be updated). It will be interesting to see if parsing the message once does improve performance, which I think there is a pretty good chance of happening; at least for UAS-like scenarios which needs to extract many headers to generate the message. I am a bit torn in that I think that one big performance advantage of SIPp is that there is no SIP stack, and thus it can generate quite a bit more load than if there were (e.g., if it were to maintain full transaction state, etc.). Also we were very interested in the transport layer and overcoming the limits in the number of different opened ports (something that you need if you want do simulate hundreds of thousands of clients). So we took a bottom-up approach with the target of having the state machines specified in the XML files. Can you post a sample of your new XML format? Also we were thinking about extending the XML files with more control and state options, things that probably David already did. You might be interested in some of the changes I recently posted that introduce the notion of numeric variables and conditional tests on those variables into the XML file, thus allowing you to do simple while loops. Overall, I think that all these changes are too radical to be integrated just as that in the SIPp trunk. Charles, you did a great job of pushing patches so far. But I think that if David also starts doing the same, it will just be too much to handle. Plus that, at least some of our changes, will kill the simplicity and ease of usage of SIPp and many current users would be upset. And I haven't even considered the new bugs that will be introduced. Yes, it is clear that a development branch or branches and some release engineering is going to be required. Charles -- Dr. Charles P. Wright Research Staff Member Network Server Systems Software IBM T.J. Watson Research Center- This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Changes for using SIPp in an implementation of ETSI TISPAN IMS Benchmark
Hi David, Verbeiren, David [EMAIL PROTECTED] wrote on 04/24/2007 04:35:10 PM: The biggest logistical problem I see with this patchset is that there has probably been quite a bit of divergence of your tree and the SIPp tree in the meantime. For example, your network changes are going to conflict with the network changes that I posted on the list a month ago. If we had more visibility into each other's efforts, hopefully there would have been peer review and less duplicated effort. I agree the network changes will be the biggest challenge initially. But just to put this back into perspective, I would like to mention that we only started coding just a little bit more than a month ago. So we haven't diverged that much. It is of course unfortunate that you've just been contributing significant changes to an area that we also completely revamped. But since we started our research on a SIPp based approach only a month ago, we couldn't really have avoided this. Just bad timing... From the depth of your changes, I had assumed that they were a much longer time in the making. Given the timelines, you are right that it seems we just ended up crossing each other in flight. One important thing to consider here is that we did not take care so far of other protocols/transports than UDP (we think our design will allow an easy integration of the others, but we did not 'port' the TCP and TLS implementations over just yet). So a combination of our efforts will hopefully be very beneficial. OK. I look forward to seeing your patch. One of the most important things that we did was to get all of the data for a single socket in a structure, and create choke point functions for sending and receiving, which is especially important for TCP and TLS (which both need to deal with congestion, partial messages, and multiple messages). I look forward to working with you towards a common code base, Charles -- Dr. Charles P. Wright Research Staff Member Network Server Systems Software IBM T.J. Watson Research Center- This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Failed to read the scenario file
Prem, If your server generates a 100 trying response, the scenario will not work; because you have no optional 100 command like: recv response=100 optional=true / You should insert this before the 401 receive command. Also, what do you mean by the statistics file counter? The screen log? Charles [EMAIL PROTECTED] wrote on 04/19/2007 02:46:31 AM: From: Premkumar V Sent: Wednesday, April 18, 2007 2:35 PM To: 'sipp-users@lists.sourceforge.net' Subject: Failed to read the scenario file Hi, These are the problems I face when I try to register a subscriber. 1. After sending the REGISTER message from the scenario.file, sipp receives 401-Unauthorised .But in the statistics file the 401 counter value remains 0(zero). 2. After receiving the 401 unauthorized or 100 trying response from the server ,sipp stops reading the scenario.file. But initial REGISTER/INVITE message is taken from the scenario.file. I have attached the scenario file below. Please kindly help me as soon as possible. Thanks Prem ?xml version=1.0 encoding=ISO-8859-1 ? !DOCTYPE scenario SYSTEM sipp.dtd scenario name=branch_client send ![CDATA[ REGISTER sip:10.20.4.242:8889 SIP/2.0 Via: SIP/2.0/UDP [local_ip]:[local_port];branch=z9hg4bk12345 From: ua1 sip:[EMAIL PROTECTED];tag=1234 To: ua1 sip:[EMAIL PROTECTED] Call-ID:123 CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:[local_port] Expires: 3600 Content-Length: 0 ]] /send recv response=401 auth=true /recv send ![CDATA[ REGISTER sip:10.20.4.242:8889 SIP/2.0 Via: SIP/2.0/UDP [local_ip]:[local_port];branch=z9hg4bk12345 From: ua1 sip:[EMAIL PROTECTED];tag=1234 To: ua1 sip:[EMAIL PROTECTED] P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E Call-ID:123 CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:[local_port] [field1] Expires: 300 Content-Length: 0 Supported: path ]] /send Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review at http://www.techmahindra.com/Disclaimer.html externally and http://tim.techmahindra.com/Disclaimer.html internally within Tech Mahindra. - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] MESSAGE method
We have been able to send MESSAGE messages, as far as I recall, there is nothing special required. Charles [EMAIL PROTECTED] wrote on 03/28/2007 11:52:43 AM: Hello everyone. Is there any special requirement to be able to send SIP packets with the MESSAGE method? I can't have it work under windows. I used the last win-install package and no cygwin. Checked with ethereal, no packet is even transmitted on the network. Registration, publish and subscribe messages are working. many thanks for your help Mathieu Davy - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] definition of cps(call per second) and call length
Hyukgeose, You should also be aware that on the UAS, there is a 4 second pause to handle retransmissions, which artificially increases the call length. Charles [EMAIL PROTECTED] wrote on 03/23/2007 04:53:28 AM: Hello Hyukgeose, cps: stands for calls per second - number of new scenarios that have been created in the last second call length: the total duration of a scenario - from first message to the last. Olivier. On 3/22/07, hyukgeose [EMAIL PROTECTED] wrote: Hi, all as you konw there are many results in screen of uac and uas. Of those I donot know what are cps and call length. Is there anyone who explains about cps and call length in detail? thanks Hyukgeose - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- HP OpenCall Software http://www.hp.com/go/opencall/ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] New Patch Set with Significant Network I/O Improvements
Carl, In general, I believe that SIPp is mostly CPU limited. You could try reducing the frequency of the clock ticks, to reduce CPU utilization, but that may cause other problems (e.g., increased burstiness). We have found that to generate high loads, you will need to use more than one machine as a UAC and UAS. There are several ways you could improve performance, if you had the motivation to improve the SIPp code itself. For example, the send_scene call could be changed such that the XML parsing breaks the message up into chunks of various types. Then, rather than strstr'ing for replacement strings, you can just switch on an integer type. There are several other examples of code that could be optimized, and if you run some profiling software you might find other low-hanging fruit. Charles [EMAIL PROTECTED] wrote on 03/23/2007 02:15:02 PM: Hey, Everyone, Now I am trying to use sipp to stress test our sip server, but the problem is that sipp seems only support several thousand (more exactly, less than 5000 thousand) concurrent tcp calls with each machine. The machine running sipp is Intel(R) Xeon(TM) CPU 3.00GHz with 2G byte memory. Does anyone have any idea about this performance issue, or how to improve sipp performace by configuring sipp? ( I only want to test tcp connection, BTW) Thanks. Carl - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Meaning of ResponseTime and ResponseTimeStDev
Juan, The values are calculated by summing the response times and call lengths in the M_counters array. To compute the average two values are needed, the sum of the response time and the number of elements that make it up. To compute the standard deviation an additional value is needed, the sum of the squares of the samples. Whenever you need to output them, the average is computed with computeMean. The standard deviation is done with computeStdev. The msToHHMMSSmmm converts the time value into a string. The counter names for CallLength are: CPT_C_AverageCallLength_Sum CPT_C_NbOfCallUsedForAverageCallLength CPT_C_AverageCallLength_Squares The _C_ means that it is cumulative. There are _PD_ and _PL_ versions are for periodic display (the screen) and periodic logging (stats file). The formula for the standard deviation is here: http://en.wikipedia.org/wiki/Standard_deviation#Rapid_calculation_methods http://upload.wikimedia.org/math/6/0/0/60036de27d964f9eb8f43add1cac001e.png All of this is in stat.cpp. Charles [EMAIL PROTECTED] wrote on 03/15/2007 04:09:46 PM: Hi, Could anyone tell me how ResponseTimeN(C)/(P) and ResponseTimeStDevN(C)/(P) values should be interpreted? How are they calculated? (at least point me to the part of the code where I can see this). Thanks a lot, Juan - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] how to avoid server mode?
Maksym, If you can control the Call-IDs that you are receiving, you can prefix them with something like foo///Call-ID and bar///Call-ID, which are both treated as having a call id of Call-ID. Otherwise, you could probably do some hacking to the get_call_id function so that it will return some constant string. Of course, this will limit you to only ever handling a single call per SIPp instance. Charles Maksym Hryhoryev [EMAIL PROTECTED] wrote on 03/15/2007 05:17:45 AM: Hello Charles, And there is no way to avoid this behaviour ? Unless the INVITEs have the same Call-ID, they will start a new scenario. I need to run sipp with scenario that begins with recv request=INVITE command but I do not need to run the scenario from beginning when another invite arrived. Is it possible? -- Best regards, Maksymmailto:[EMAIL PROTECTED] - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] New Patch Set with Significant Network I/O Improvements
Olivier, That sounds like a good idea, there have been lots of bug fixes, etc. since then and getting the new code into the hands of more users would be great. The extensive changes are limited to iomodel.diff, the rest of them are pretty simple. Unfortunately, there was no good way to break up the iomodel patch into smaller ones (once you start changing one thing there is a large cascade). The other thing that you might want to skip in rc9 is the warnings.diff change to the Makefile which adds -Wall -Werror. This is good for developers, because it forces us to do things a little bit more cleanly and can catch bug-a-boos like incorrect printf formats, functions w/ no returns etc. However, depending on the exact system the user compiles it on, there could be warnings that prevent SIPp from compiling. I think a good middle-ground position would be to apply the warning fixes, but not the Makefile change in rc9, but put the Makefile change in SVN after rc9. Charles Olivier Jacques [EMAIL PROTECTED] wrote on 03/14/2007 06:29:54 PM: Charles, we are looking into those. What I will do is do an rc9 without those changes that are pretty extensive (rc8 is from December 06). That will give us a bit of time to stabilize. Thanks! Olivier. On 3/14/07, Charles P Wright [EMAIL PROTECTED] wrote: Hello all, I've attached a new patch set with the following patches: - vgfix.diff Various valgrind fixes. - warnings.diff Allow the code to compile with -Wall -Werror on Linux. - doublelost.diff Allow loss percentages less than 1 and also a global command line option to specify that packets should be lost at a given percentage. - usedrtds.diff Only include RTDs that are actually used in the CSV output. - micrortt.diff Use RTDs that are precise to the microsecond in -trace_rtt, and improve the consistency between trace_rtt and the averages. - iomodel.diff Completely reworked the network I/O subsystem so that all of the code goes through a single read and single write function. This ensures that no partial writes can get mixed, and eliminates TCP read deadlocks. The code is also cleaner as all of the information related to a given socket is stored in one structure. - clockupdate.diff Update the clock_tick more frequently so that we have a higher timer and statistics resolution. The biggest thing here is the new I/O model (actually the biggest patch that we have submitted to date). I have basically reworked the entire read and write paths for network sockets, and encapsulated all of the information SIPp needs about the socket into a single structure (sipp_socket). The same primitive functions are used regardless of the socket type and there are fewer layers that must handle various error conditions (e.g. congestion). Aside from a general cleanup, this addresses two important problems: (1) We have observed that under TCP congestion SIPp was truncating packets and subsequent packets would be sent out before the partial message went through. (2) SIPp using TCP can not talk to another SIPp instance using TCP, because they would deadlock. The reason is that SIPp used to block until a whole message is read and both peers could have sent a partial message due to congestion. This results in both of the SIPp instances waiting to read, and never completing the partially sent message. I have tested basic UDP (w/ and w/o multi sockets), TCP (w/ and w/o multi sockets), TLS (w/ and w/o multi sockets), 3PCC, and per-IP sockets. I have not been able to test compression (there are no public plugins I am aware of) and 3PCC extended (there are no publically available scenarios I could see). Aside from correcting the flaws that I mentioned and cleaning up the code, this should improve the extensibility of the code for two reason: (1) New information can easily be stuck in the socket structure. (2) There is an accurate mapping of calls to sockets (and an associated reference count) (3) There is less reliance on global variables for the network primitives. For example, transport and ipv6 are stored in the socket structures. This should (theoretically) make it easier to mix TCP and UDP or IPv4 and IPv6 in the future. Of course, if you have any questions about any of these patches, I'll be glad to answer them, Charles -- Dr. Charles P. Wright Research Staff Member Network Server Systems Software IBM T.J. Watson Research Center - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV
Re: [Sipp-users] Having an optional recv before send
Harsimran, Try something along the lines of the following: recv request=BYE timeout=timeout ontimeout=1 / label id=1 / send ... BYE ... /send Charles [EMAIL PROTECTED] wrote on 03/14/2007 02:09:34 AM: Hi How can I design a scenario to have a optional recv before a send sequence. What I want my scenario is to wait for a BYE for some time if it doesnt gets it then the scenario should send a BYE. Is there any way of doing that. Thanks Harsimran Singh - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] how to avoid server mode?
Unless the INVITEs have the same Call-ID, they will start a new scenario. Charles [EMAIL PROTECTED] wrote on 03/14/2007 04:57:18 PM: Hello sipp-users, I need to run sipp with scenario that begins with recv request=INVITE command but I do not need to run the scenario from beginning when another invite arrived. Is it possible? -- Best regards, Maksym mailto:[EMAIL PROTECTED] - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Question on call scenario
Anil, Make sure that your REGISTER and INVITE have the same Call-ID. If you need to have two different call-id's you can use /// to prefix them with something. For example: Call-ID: FOO///[EMAIL PROTECTED] and Call-ID: BAR///[EMAIL PROTECTED] should be treated as the same Call-ID by SIPp. Charles [EMAIL PROTECTED] wrote on 03/13/2007 10:22:38 AM: Hi, I am running a single call that has the following call scenario: SIPa SIPb --REGISTER ---200 OK -- -INVITE -180 Ringing -200 OK--- When I run the scenario with sipa.xml and sipb.xml on 2 different machines, SIPb receives REGISTER and sends 200 OK. SIPa sends INVITE to SIPb, but SIPp on SIPb machine prints the following log: Aborting call on unexpected message for Call-ID '109771157-2561': while expecting 'REGISTER', received 'INVITE'.. When SIPb is expecting INVITE according to the call flow(after sending 200 OK), then why is the application throwing error on REGISTER. Please help me on this scenario. (This is just one call, not in retransmission mode) This is on SIPp v1.1-TLS-PCAP, version 20061124 version. XML file SIPb.xml == ?xml version=1.0 encoding=ISO-8859-1 ? !DOCTYPE scenario SYSTEM sipp.dtd scenario name=SIPa to SIPb recv request=REGISTER crlf=true /recv send ![CDATA[ SIP/2.0 200 OK /send recv request=INVITE crlf=true /recv send ![CDATA[ INVITE... /send == - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Including the random time from a pause in a logfile
You can put a NOP before and after the pause with a start_rtd and a rtd, and then use -trace_rtt. For example: nop start_rtd=2 / pause normal=true mean=100 stdev=10 / nop rtd=2 / If you look at the scenario_pid_rtt.csv you can pick out the RTTs with a Rtd_no of 2. Charles [EMAIL PROTECTED] wrote on 03/12/2007 05:06:45 AM: If I do a pause with a min and max, is there any way of taking that value and placing it in a logfile (As if I wanted to have a logfile of all the random pauses)... -- Rizwan Kassim Software and Systems Engineer, uWink Interactive Bistro http://www.geekymedia.com If you have a problem and you think awk(1) is the solution, then you have two problems. -David Tilbrook - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Updated Average and Standard Deviation Code
(2) The update_nb option seems to take great care to avoid calling getmilliseconds() on every loop. Is there a particular system that this call is very expensive on? Olivier, I've done a quick test on Linux and Windows and found that it is quite fast (395/390 nanoseconds), certainly compared to some of the heavy lifting that the main processing has to do. It seems that this is cheap enough that removing the up_nb option, or at least making the default behavior to get the time more often (e.g., every loop and call to call::run()) would be a good idea. If you agree, I'll work up a patch to that effect. Charles- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Retransmission Counter Fix
Hello, This patch corrects a couple of bugs in received transmission handling: (1) If a send is unsuccessful, the retransmission should still be counted as received. (2) Cookies for optional messages should also be recorded so that the has comparison works. (3) If a message matches last_recv_hash, then we should increment the retransmission counter. Charles -- Dr. Charles P. Wright Research Staff Member Network Server Systems Software IBM T.J. Watson Research Center retranshash.diff Description: Binary data - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] reinvite with UAS and peer_tag_param issue
Paul,
From a quick look at the code, it seems that the peer tag is only picked
up out of responses and not replies, and in your scenario there are no
received replies before you use the peer_tag_param.
You should take a look at this bit of code:
/* It is a response: update peer_tag */
ptr = get_peer_tag(msg);
if (ptr) {
if(strlen(ptr) (MAX_HEADER_LEN - 1)) {
ERROR(Peer tag too long. Change MAX_HEADER_LEN and recompile
sipp);
}
if(peer_tag) { free(peer_tag); }
peer_tag = strdup(ptr);
if (!peer_tag) {
ERROR(Out of memory allocating peer tag.);
}
}
In call.cpp and possibly move it outside of the /* Is it a response ? */
statement.
I hope these hints are helpful.
Charles
Paul Antinori (pantinor) [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
03/06/2007 10:25 AM
To
sipp-users@lists.sourceforge.net
cc
Subject
[Sipp-users] reinvite with UAS and peer_tag_param issue
Hi,
I am sending a re-invite with a UAS script in SIPp and am having trouble
with the [peer_tag_param] not getting populated on the To header.
Is there anything I am missing? See my script below.
Thanks for any help,
Paul
recv request=INVITE/recv
send
![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: sip:[local_ip]:[local_port];transport=[transport]
Content-Length: 0
]]
/send
send retrans=500
![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: sip:[local_ip]:[local_port];transport=[transport]
Content-Type: application/sdp
Content-Length: [len]
]]
/send
recv request=ACK rtd=true crlf=true/recv
!-- some 10 second talk time before putting caller on hold --
pause milliseconds=1 crlf=true/
send retrans=500
![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From:
sip:[EMAIL PROTECTED]:[local_port];transport=[transport];tag=[call_number]
To: sip:[EMAIL PROTECTED]:[remote_port][peer_tag_param]
[last_Call-ID:]
CSeq: 101 INVITE
Remote-Party-ID:
sip:[EMAIL PROTECTED];party=calling;screen=no;privacy=off
Contact: sip:[local_ip]:[local_port];transport=[transport]
Paul Antinori
Software Engineering
CCBU - Voice Technology Group
[EMAIL PROTECTED]
Phone :978-936-1798
Cisco Systems, Inc.
500 Beaver Brook Road
Boxborough, MA 01719
www.cisco.com
-
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[Sipp-users] Understanding main_socket vs. tcp_multiplex
Can someone explain the difference between the main_socket and the tcp_multiplex socket? It seems that the TCP multiplex socket should only be used if we are using -t t1 (otherwise each call gets its own socket). But, why are things sent over the tcp_multiplex socket instead of just the main_socket? Charles- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Route and Record-Route...
Peter/Olivier,
I think instead of adding a start_line attribute, that behavior should be
the default (as most people probably expect it to work that way).
Charles
Olivier Jacques [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
02/14/2007 02:15 PM
To
Peter Higginson [EMAIL PROTECTED]
cc
sipp-users@lists.sourceforge.net, Thomas Rosenblatt
[EMAIL PROTECTED]
Subject
Re: [Sipp-users] Route and Record-Route...
Peter,
thanks for the heads up, I'll have a look tonight.
Olivier.
On 2/13/07, Peter Higginson [EMAIL PROTECTED] wrote:
I did submit these suggested changes a long time ago. Either occurrence or
start_line would do what you want. (I cannot remember why they did not get
put in the main version - there may have been a good reason.)
Peter
Peter Higginson
Newport Networks Ltd,
Direct line 01494 470694
http://www.newport-networks.com/
=
From: Peter Higginson [mailto: [EMAIL PROTECTED]
Sent: 26 August 2005 10:07
To: 'Siddharth Angrish'; 'sipp-users@lists.sourceforge.net'
Subject: RE: [Sipp-users] Query regarding sipp regexp
Some changes we have made in our local copy to meet this requirement are
to
add the following functionality to regexp for the hdr matching case:
1) case_indep=true to look for a header ignoring case
2) occurrence=n to find the nth occurrence of a header
3) start_line=true to look only at start of line
The occurrence option will allow you to set different variables to each of
the headers you are looking for.
I have also fixed a bug which might crash SIPp for packets longer than
1024
bytes (this may be already incorporated). Note that the current hdr match
looks anywhere in the message and I have left this as the default (with
start of line as the option).
The code changes are below (I've done them based on the version we use,
but
I think they match up with the current versions):
In scenario.cpp, change this section of code:
tmpAction.setVarType(CAction::E_VT_REGEXP);
tmpAction.setActionType(CAction::E_AT_ASSIGN_FROM_REGEXP);
if(ptr = xp_get_value((char *)search_in)){
if(!strcmp(ptr, (char *)msg)) {
tmpAction.setLookingPlace(CAction::E_LP_MSG);
tmpAction.setLookingChar(NULL);
} else if (!strcmp(ptr, (char *)hdr)) {
if(ptr = xp_get_value((char *)header)) {
if(strlen(ptr) 0) {
tmpAction.setLookingPlace(CAction::E_LP_HDR);
tmpAction.setLookingChar(ptr);
} else {
tmpAction.setLookingPlace(CAction::E_LP_MSG);
tmpAction.setLookingChar(NULL);
}
To:
tmpAction.setVarType(CAction::E_VT_REGEXP);
tmpAction.setActionType (CAction::E_AT_ASSIGN_FROM_REGEXP);
// warning - although these are detected for both msg and hdr
// they are only implemented for search_in=hdr
if ( 0 != ( ptr = xp_get_value((char *)case_indep) )
0 == strcmp(ptr, true)) tmpAction.setCaseIndep(true);
else tmpAction.setCaseIndep(false);
if ( 0 != ( ptr = xp_get_value((char *)start_line) )
0 == strcmp(ptr, true)) tmpAction.setHeadersOnly(true);
else tmpAction.setHeadersOnly(false);
if ( 0 != ( ptr = xp_get_value((char *)search_in) ) ) {
tmpAction.setOccurrence(1);
if ( 0 == strcmp(ptr, (char *)msg) ) {
tmpAction.setLookingPlace(CAction::E_LP_MSG);
tmpAction.setLookingChar (NULL);
} else if (!strcmp(ptr, (char *)hdr)) {
if ( 0 != ( ptr = xp_get_value((char *)header) ) ) {
if ( 0 strlen(ptr) ) {
tmpAction.setLookingPlace(CAction::E_LP_HDR);
tmpAction.setLookingChar(ptr);
if (0 != (ptr = xp_get_value((char *)occurrence))) {
tmpAction.setOccurrence (atol(ptr));
}
} else {
tmpAction.setLookingPlace(CAction::E_LP_MSG);
tmpAction.setLookingChar(NULL);
}
In actions.hpp, after
char* getLookingChar();
Add:
bool getCaseIndep();
intgetOccurrence();
bool getHeadersOnly();
After:
void setAction (CActionP_action);
Add:
void setCaseIndep(bool P_value);
void setOccurrence (intP_value);
void setHeadersOnly (bool P_value);
and after:
intM_varId;
Add:
bool M_caseIndep;
intM_occurrence;
bool M_headersOnly;
In actions.cpp, after:
char* CAction::getLookingChar() { return(M_lookingChar); }
Add:
bool CAction::getCaseIndep(){
Re: [Sipp-users] call-id revisited in server mode
One option that may work for you is to use two separate SIPp instances (one for registration and another for the UAS behavior). Charles [EMAIL PROTECTED] wrote on 02/07/2007 10:41:38 AM: First I want to thank the maintainers for an incredibly useful tool and a good job! Now my question: I have searched the archives and the documentation with no luck to my specific question. My scenario looks like this: -- REGISTER 0 0 0 -- 2000 0 -- INVITE 0 0 0 -- 2000 0 -- ACK0 0 0 -- BYE0 0 0 -- 2000 0 Now the problem I have is that I'm testing a sip user agent that registers with one call-id and sends the invite with another. This is often the case in real life because the register is not the call session. Is there any way for me to get this scenario to work with SIPp? If I ask a dumb question, feel free to shoot me ;-) Best regards Niklas Fondberg - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier. Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier. Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] call-id revisited in server mode
No, unfortunately not. You may be able to get around this though by using optional and next. Try putting REGISTER as the first message, but mark it optional with a next of the 200 reply at the end of the scenario. Also make the 200 reply to the bye jump to the end of the scenario. Something like: recv optional=true method=REGISTER next=1 / recv method=INVITE / ... rest of call ... send next=2 (for bye) 200 ... /send label id=1 / send retrans=500 200 ... /send label id=2 / Charles Niklas Fondberg [EMAIL PROTECTED] wrote on 02/07/2007 10:55:15 AM: Thanks for the quick reply. Can this be done using the same ip:port? Niklas On Wed, 2007-02-07 at 10:48 -0500, Charles P Wright wrote: One option that may work for you is to use two separate SIPp instances (one for registration and another for the UAS behavior). Charles [EMAIL PROTECTED] wrote on 02/07/2007 10:41:38 AM: First I want to thank the maintainers for an incredibly useful tool and a good job! Now my question: I have searched the archives and the documentation with no luck to my specific question. My scenario looks like this: -- REGISTER 0 0 0 -- 2000 0 -- INVITE 0 0 0 -- 2000 0 -- ACK0 0 0 -- BYE0 0 0 -- 2000 0 Now the problem I have is that I'm testing a sip user agent that registers with one call-id and sends the invite with another. This is often the case in real life because the register is not the call session. Is there any way for me to get this scenario to work with SIPp? If I ask a dumb question, feel free to shoot me ;-) Best regards Niklas Fondberg - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier. Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier. Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] TCP problems
What release of SIPp are you using? The latest (1.1-rc8) has lots of TCP fixes, which should hopefully solve your problems. Charles [EMAIL PROTECTED] wrote on 01/09/2007 01:26:53 PM: Hi, I am using SIPp and SER for some performance study and get several errors when using TCP protocol. From the packet trace, I notice when TCP publicize zero window at SIPp or SER side, SIP message will get truncated or several partial or whole SIP message get combined in one TCP packet, which leads to error. I haven?t checked the code yet, but if somebody already knows this off the head, please clarify me of the following questions. 1) When SIPp receives a truncated SIP message, how does it handle the partial message? 2) When SIPp receives a packet that has one partial SIP message combined with another partial SIP message, how does it handle the packet? I joined the alias recently, so if the alias already had discussions on TCP stream behavior and its effects on SIP implementation, could somebody point the link to me? Thanks, Joy __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Authentication issues in sipp 2007-01-03 snapshot
The problem seems to be this memset line at call.cpp:2196 (or thereabouts): memset(my_auth_pass,0,KEYWORD_SIZE); key = getKeywordParam(src, password=, my_auth_pass); If you remove that line, the authentication from the command line should be used. Charles [EMAIL PROTECTED] wrote on 01/09/2007 05:25:34 AM: Hi Jason! I had the same problem with snapshot 2007-01-02. Using [authentication username=foo password=bar] works, but the command line parameters didi not worked. regards klaus Jason Wever wrote: Hi All, Seeing some interesting behavior in the SIPp 2007-01-03 snapshot on Linux and wondering if anyone else has seen this. I have a REGISTER scenario that simply registers a user and then logs out. Originally, I had the scenario checking for receipt of a 401 with auth=true and then sending another registration with the authentication information. I had setup SIPp so that the authentication line in the scenario looked like [authentication username=foo] and was passing the password via the command line. This worked in the 2006-08-29 snapshot, but using later snapshots (like 2006-12-08 or 2007-01-03) caused a 401 to be received to the second register rather than a 200 OK. However, I did notice if I put the password information in the authentication keyword (for example [authentication username=foo password=bar] that the newer SIPp snapshots can now register like the 2006-08-29 snapshot did. As the password option on the SIPp command line still shows up in help (sipp -h), is it still being honored or might this be a bug? Is anyone else seeing this? Thanks, -- Klaus Darilion nic.at - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] TCP problems
SIPp will read a message until it finds a \r\n\r\n, at which point it knows that it received the headers. Next it finds the Content-Length header in the message, and if it exists reads the number of bytes specified in the content length header. A message is not complete until the full number of bytes have been read from the socket. A message can not be a partial message, unless an error occurred on the socket. What I did is insert a bit of buffering, so that SIPp will suck up the available bytes in the TCP buffer into a local buffer in one system call. The next time SIPp needs to read one or more bytes, it looks in the local buffer first; avoiding the system call. This does not change any of the logic for determining if something is partial or complete. Charles Katty Xiong [EMAIL PROTECTED] wrote on 01/09/2007 03:49:55 PM: Hi Charles, I see you have made a lot of changes to TCP handling in SIPp. Could you explain how SIPp decides if a SIP message is complete or partial? Since I suspect the problem I am seeing could be caused by SIP message compatibility issue between SER and SIPp. thanks, Joy --- Charles P Wright [EMAIL PROTECTED] wrote: What release of SIPp are you using? The latest (1.1-rc8) has lots of TCP fixes, which should hopefully solve your problems. Charles [EMAIL PROTECTED] wrote on 01/09/2007 01:26:53 PM: Hi, I am using SIPp and SER for some performance study and get several errors when using TCP protocol. From the packet trace, I notice when TCP publicize zero window at SIPp or SER side, SIP message will get truncated or several partial or whole SIP message get combined in one TCP packet, which leads to error. I haven?t checked the code yet, but if somebody already knows this off the head, please clarify me of the following questions. 1) When SIPp receives a truncated SIP message, how does it handle the partial message? 2) When SIPp receives a packet that has one partial SIP message combined with another partial SIP message, how does it handle the packet? I joined the alias recently, so if the alias already had discussions on TCP stream behavior and its effects on SIP implementation, could somebody point the link to me? Thanks, Joy __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIP -d command-line option
Maksym, The duration is specified in milliseconds, so you should use -d 5000 for a five second pause. Charles [EMAIL PROTECTED] wrote on 12/25/2006 05:13:03 AM: Hello sipp-users, I run SIP client with -d command-line option. sipp ... -d 5 ... XML scenario contains pause/ string (without milliseconds option). expect to get a 5 seconds delay in the call. But no pause appears. Question: option only? -- Best regards, mailto:[EMAIL PROTECTED] - Take Surveys. Earn Cash. Influence the Future of IT SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp 1.1rc8 released
Olivier, Replace line 3535: *((int *)option-data) /= 1000; with this: *((double *)option-data) /= 1000; And Amit's problem should be solved.) FYI, the timeoption.diff patch that I sent yesterday provides much more consistent and flexible control over these options (e.g., -rp 5s or -rp 10ms or even -rp 0.5s). Charles Jacques, Olivier (PDE IT Test) olivier.jacques@ To hp.com Amit On [EMAIL PROTECTED], Sent by: sipp-users@lists.sourceforge.net sipp-users-bounce cc [EMAIL PROTECTED] ge.netSubject Re: [Sipp-users] SIPp 1.1rc8 released 12/22/2006 09:25 AM Amit, -Original Message- From: Amit On [mailto:[EMAIL PROTECTED] Sent: Thursday, December 21, 2006 13:11 sipp-users@lists.sourceforge.net; SIPp 1.1rc8 released and we found that there is a and in the documentation MilliSeconds. But in the current You can try it out with this command and see that it says that it will 5 Scenario's every 1000 Seconds instead of every 1000 milliseconds: uac 127.0.0.1 -r 5 -rp 1000 [ M ] +972(0)525222810 [ T ] +972(0)48142232 [ F ] +972(0)48 550 [ W ] www.followap.com / www.neustar.biz From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacques, Sent: Wednesday, December 20, 2006 8:56 PM sipp-users@lists.sourceforge.net released never released), many new features and fixes main one being: a factor of pauses emulate real user behaviors platforms increase automatically the call rate at specific Fixes for AKA (IMS) authentication Full release notes: http://sourceforge.net/project/shownotes.php?group_id=104305release_id=: 472717 http://sourceforge.net/project/showfiles.php?group_id=104305package_id=: 119322release_id=472717 This is the occasion to warmly thank all the contributors that to this release, with a special thanks to Charles P. Wright research for a huge set of new features and improvements. that some fixes and enhancements didn't make their way especially thinking of FreeBSD patches that I didn't had pre-post scenarios that didn't started as we wanted and we will to continue integrating those as fast as we are ahead of us, stay tuned! See you in 2007, amazing for a SIP test http://sourceforge.net/project/stats/detail.php?group_id=104305ugn=sipp type=prdownloadmode=alltimepackage_id=0 Take Surveys. Earn Cash. Influence the Future of IT Join Techsay panel and you'll get the chance to share your business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDE V list https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Techsay panel and you'll get the chance to share your business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ list https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Some Minor Changes New Option Handling
Olivier, Thanks for applying this patch set. I hope that things like the options table will allow more error checking and consistency to creep into the code over time. One question that I had about the XML parser is if there is a way to iterate over the attributes in an element? I didn't see one, but wanted to make sure I was not missing anything. It would be nice to be able to have a simple syntax checker that makes sure there aren't any extra misspelled attributes (e.g., I added a repeat_rtd attribute, but inverted it to rtd_repeat while testing). I posted a new patch set this morning (but it has not yet appeared on the list) against 12/19, please ignore that patch set and instead use this one which is against the 12/20 release. The usersoption.diff patch actually fixes a bug in the original set that I sent against the 12/8 release, in which I forgot to include the -users option in the table because it was not included in the help message. - makefile.diff Include an EXTRAENDLIBS keyword so that libraries can be appended to the list after SSL. Fail when parsing a scenario that enables authentication if SSL is not enabled. - pcapcheck.diff Fail when parsing a scenario that has pcap if pcap is not enabled. For options that take a time allow them to be specified using seconds or milliseconds. This lets you have more precise statistics intervals, or just a more convenient way to specify times. Use the stat_delimiter for the trace_rtt option, and also include the number that is being reported. Finally, if a scenario loops back on itself you can enable repeated RTD calculations with the repeat_rtd XML element. Add the -users option to the table. Charles (See attached file: sipp-patches-2006-12-20-ibm1.tar.gz) Olivier Jacques [EMAIL PROTECTED] wrote on 12/20/2006 10:55:37 AM: Charles P Wright wrote: patches to the 2006-12-08 release. Charles, I have checked all the changes in. http://sipp.sourceforge.net/snapshots/sipp.2006-12-20.tar.gz Thanks a lot, again, -- Olivier sipp-patches-2006-12-20-ibm1.tar.gz Description: Binary data - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIP URL and IPv6
You can use \x to have a hex-encoded character. Try doing: INVITE sip:[EMAIL PROTECTED]::214:4fff:fe22:d312] SIP/2.0 This should prevent SIPp from treating the IPV6 address as a keyword. Charles [EMAIL PROTECTED] wrote on 12/13/2006 08:08:05 AM: Hi Friends, I am trying to use an IPv6 address for a SIP URL in a scenario file. “INVITE sip:[EMAIL PROTECTED]::214:4fff:fe22:d312] SIP/2.0” But SIPp tries to interpret the IPv6 address as a keyword and throws the following error. “Unsupported keyword ‘fe80::214:4fff:fe22:d312’ in xml scenario file.” How can I overcome this issue? Also is there any way to escape the keywords in a scenario file? Thanks, Vasu - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Some Minor Changes New Option Handling
Hello All, I have attached a series of patches to the 2006-12-08 release. The largest one is optionhandling.diff, which is a pretty serious and invasive overhaul of the SIPp option handling. Instead of adding an additional if statement for each option, a table entry is added. Each table entry contains: (1) The name of the option. For example: m (2) Help text. For example: Stop the test and exit when 'calls' calls are processed The help function will automatically word-wrap the text to 80 characters, and handles a single level of bullets using a - sign. (3) The type. For example: SIPP_OPTION_INT (4) A pointer: For example: stop_after. The hope is that this will make the code cleaner, and ensure that help text always gets updated when a new option is introduced. - retransoption.diff This adds the ability to specify the maximum retransmission for invites and non-invite messages. Count messages before actions are performed. Use get_ functions for numbers in the scenario, so that more invalid scenarios will be caught. Use a table driven option parsing architecture. This makes it easier to add new options, and should increase error handling of invalid options. Also use the option table to automatically generate the help message. Charles -- Dr. Charles P. Wright Software (See attached file: sipp-patches-2006-12-08-ibm1.tar.gz) sipp-patches-2006-12-08-ibm1.tar.gz Description: Binary data - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Minor Enhancement and Fix
Hello All, I've attached two patches that make some minor enhancements to the latest SIPp (11/24/06). - retransoption.diff This adds the ability to specify the maximum retransmission for invites and non-invite messages. - countbeforeaction.diff Count messages before actions are performed. The second is important, because if you have a scenario action that rejects the call, the message would not be counted as received on the screen, towards response time distributions, or other counters. Additionally, the action would be performed even if there was a simulated loss. Charles -- Dr. Charles P. Wright Software (See attached file: retransoption.diff)(See attached file: countbeforeaction.diff) retransoption.diff Description: Binary data countbeforeaction.diff Description: Binary data - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Some Fixes and Enhancements
Hello, I have attached a few patches with some enhancements and a few bug fixes to my previous patches. The major enhancement is that TCP reads are no longer octet-by-octet at the system call level. The bug fixes are for TCP partial message handling and also pcap playing. I haven't actually used pcap playing before, so don't know how to properly test it, but this fix does cause the socket to be opened and Wireshark shows RTP packets flowing. - partialtcpmessage.diff Fix handling of partial TCP messages by entering congested state, otherwise partial messages end up getting corrupted. - pollset.diff Add buffers in front of the TCP sockets, so that octet-by-octet reads only require a function call and not a system call. This improved TCP performance by a factor of four during my tests. - mpcycle.diff If the media port is not available cycle upwards until a free one is found (as is done for the SIP and control ports). - csvdelim.diff Allow alternative strings to delimit the statistics file, if set to a comma, this makes it more convenient to open the files in Excel. - valgrind.diff A few things that I found by running SIPp through valgrind. - reconnect.diff Better control reconnection behavior (the pause and whether or not to close all of the calls). - fixpcap.diff This is a two line fix to pcap playing that was broken in the course of my previous patch set. Charles (See attached file: sipp.2006-11-08-ibm1.tar.gz) sipp.2006-11-08-ibm1.tar.gz Description: Binary data - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] TCP Error Handling Questions
Olivier, I've included a TCP modification in the latest patch set that I just sent to the list. It corrects three deficiencies: 1. If EAGAIN is returned, then return -1 from the send function so that the call knows that this message was not sent successfully and will try again. 2. Allow the EAGAIN handling to be performed more than once per execution. The EAGAIN condition is actually a normal condition on a Unix machine when there is congestion, so rather than only doing the throttling one time, the throttling is performed whenever congestion is encountered. 3. Calls where the first message encountered congestion were deleted, but the statistics did not mark them as failed. Instead of deleting the call, the patch lets it try again. I have tested this with the t1 transport option, and it does not noticeably increase the call rate that I can sustain, but it does reduce the number of failed calls at higher loads. Charles Olivier Jacques [EMAIL PROTECTED] com To Bruno, Guerin (NonHP : 10/20/2006 09:58 AtosOrigin) [EMAIL PROTECTED], AMCharles P Wright/Watson/[EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net Subject Re: [Sipp-users] TCP Error Handling Questions Bruno, thanks for the analysis. Charles, as Bruno pointed, we are not very comfortable with the way TCP congestion is currently implemented. Well, there are several outstanding issues wrt TCP: - TCP reads are octet per octet (far from being to say the least!) - There was a message from Ajay Gupta, back in May 25th. There are 2 fixes for TCP that looks useful I will integrate that asap. - Are you looking to work on something else? Olivier. On 10/20/06, Bruno, Guerin (NonHP : AtosOrigin) [EMAIL PROTECTED] wrote: Hello Concerning TCP congestion, here are some answers: 1) If a message is truncated, the rest of the message (not sent) is sent as soon as is possible, i.e. just after a received on the socket and before any other send. 2) The purpose of ctrlEWGlobal is to trace than a problem occurs once (independantly of the socket descriptor). The philosophy behind this is : -if a problem occurs, a chance is let to solve it. -if a second problem occurs (even on a different socket), it is because the system does not support such traffic, so it is not useful to treat the problem again. In the implementation, if a first pb occurs, the value of ctrlEWGlobal is set to true. If traffic is in mono socket, 'ctrlEW' is set to true. If traffic is in multi socket a flag, attached to the socket descriptor (poll_flag_write) is set to true. No more sent are done till 'ctrlEW' or the flag is set to false. If a message is received on the socket, it is supposed that the pb is solve. 'ctrlEW' or the flag is set to false. SIPp re-starts to send message. If a new problem occurs (even on a different socket in multi socket traffic), nothing is done to stop the traffic. 'ctrlEWGlobal' prevents to enable the protection (no send) again. SIPp continue to trying to send and receive message. This implementation is probably not the best one. 3) 'ctrlEW' is used in mono socket traffic to trace if a problem (EAGAIN or EWOULDBLOCK) occurs during the last send. If a problem occurs 'ctrlEW' is set to true, no more send are done. If a message is received, it is supposed that network problems are solve, so 'ctrlEW' is set to true. SIPp will try to send message again. Concerning incorrect behavior when the TCP window closed and send returned EWOULDBLOCK or EAGAIN problem, no investigation has been done yet. Regards Bruno GUERIN -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Charles P Wright Envoyé : mardi 17 octobre 2006 23:12 À : sipp-users@lists.sourceforge.net Objet : [Sipp-users] TCP Error Handling Questions Hello
[Sipp-users] TCP Error Handling Questions
Hello, I have a question regarding TCP support under congestion in the latest SIPp releases. Previously, there was discussion on the mailing list that pointed out incorrect behavior when the TCP window closed and send returned EWOULDBLOCK or EAGAIN. I looked at the send_message function, and have the following questions: 1. Why are partial messages treated specially? Why not use the same code for messages that were not sent at all (an extreme case of being truncated)? 2. What is the purpose of ctrlEWGlobal? It seems that it causes the code to be executed only once, because I never see a false value assigned to it. 3. What is the purpose of ctrlEW? Thanks, Charles - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
