Re: [Sipp-users] Dead call (successful) error
Hi, according to your screenshot, I assume that you have a syntax-error in your scenario-defintion. br Michael On 25.06.2013 22:31, Max Magee wrote: -- Scenario Screen [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 5060 0.20 s2 SERVER:5060(TCP) Call limit reached (-m 2), 0.000 s period 0 ms scheduler resolution 0 calls (limit 30) Peak was 1 calls, after 0 s 1 Running, 3 Paused, 0 Woken up 1 dead call msg (discarded)0 out-of-call msg (discarded) 2 open sockets Messages Retrans Timeout Unexpected-Msg *REFER -- 2 0 * -- Test Terminated -- Michael Hirschbichler, Mag. Dipl.-Ing. Institute of Telecommunications Vienna University of Technology A-1040 Wien, Favoritenstr. 9-11/388 Phone: +43 1 58801 38846 -- This SF.net email is sponsored by Windows: Build for Windows Store. http://p.sf.net/sfu/windows-dev2dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] SIPp-wish :)
Hi all, I am currently developing another SIPp-XML-scenario and I have one *wish* for future functionalities: Today, if I want to analyse the response codes of e.g. a failed call, it looks this way: recv response=400 optional=true next=2 /recv recv response=401 optional=true next=2 /recv recv response=403 optional=true next=2 /recv recv response=404 optional=true next=2 /recv recv response=407 optional=true next=2 /recv It is clear, that the XML is quite unreadable and the responsecodes are still incomplete - if a 402 is returned, this responsecode is not catched. Now my proposal: it would be great if we can aggregate multiple respnses into one recv-statement: recv response=4?? optional=true next=2 /recv or recv response=40? optional=true next=2 /recv recv response=41? optional=true next=3 /recv or even recv response=(400|401|402) optional=true next=2 /recv This would be a massive impromevent for creating generic and readable XML-files BR Michael -- Everyone hates slow websites. So do we. Make your web apps faster with AppDynamics Download AppDynamics Lite for free today: http://p.sf.net/sfu/appdyn_d2d_mar ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Simple to configure SIP Proxy
Why needing two proxies? Which additional functionality should a proxy offer compared to SIPp? SIPp --- *SUT* --- SIPp br Michael On 08.03.2013 16:15, Santosh Reddy wrote: Hi, I need to use a proxy for testing my system under test. The scenario will be SIPp --- Proxy --- *SUT *--- Proxy --- SIPp Can you please put your thoughts on proxies which you have used based on below points. My requirements from Proxy are: * Should be simple to configure * Should not modify any headers * Should transparently pass additional or custom headers * Should not modify SDP * Routing should be configurable based on userpart, domain * Support for Registration Thanks Regards, Santosh Reddy. -- Symantec Endpoint Protection 12 positioned as A LEADER in The Forrester Wave(TM): Endpoint Security, Q1 2013 and remains a good choice in the endpoint security space. For insight on selecting the right partner to tackle endpoint security challenges, access the full report. http://p.sf.net/sfu/symantec-dev2dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Michael Hirschbichler, Mag. Dipl.-Ing. Institute of Telecommunications Vienna University of Technology A-1040 Wien, Favoritenstr. 9-11/388 Phone: +43 1 58801 38846 -- Symantec Endpoint Protection 12 positioned as A LEADER in The Forrester Wave(TM): Endpoint Security, Q1 2013 and remains a good choice in the endpoint security space. For insight on selecting the right partner to tackle endpoint security challenges, access the full report. http://p.sf.net/sfu/symantec-dev2dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp 3.4 roadmap
Hi Rob, mentioning the error-log: can you take a look at the time-formatting of the -trace_msg logs? Currently, they are formatted as 2013-02-16 23:40:21:589.216 which is incorrect. The correct (parseable) formatting should (IMHO) be 2013-02-16 23:40:21.589216 (as the ms und us are fractions of a second). Additionally, the leading zeros of the microseconds are removed. E.g. instead of 2013-02-16 23:40:21:573.012 SIPp formats it as 2013-02-16 23:40:21:573.12 The -trace_shortmsg-logs are not affected, as these logs contain the unix-timestamps 1361058600.182012. Well, but in this context: the timestamps of the trace_msg and trace_shortmsg differ. The reason for this behaviour is the point of time when the timestamp is stored. This is done in different lines in the code resulting in different timestamps. Thanks for your contribution! BR Michael Am 15.02.2013 23:37, schrieb Rob Day: Hi all, I've uploaded the SIPp 3.3 packages, and now that that release is complete, I'm planning what to include in SIPp 3.4. My current thoughts are: * Review and potential inclusion of patches at https://sourceforge.net/p/sipp/patches/milestone/v3.4/ * Fixes for bugs at https://sourceforge.net/p/sipp/bugs/milestone/v3.4/ (excepting perhaps the RTP threading issues) * Port from OpenSSL to GnuTLS (for licensing reasons) * Add a ./configure option to enable the GNU Scientific Libraries * Logging improvements: * Short error log - every minute (or other time period determined by -fd), print out a list of SIP error codes received (e.g. timestamp: 503 404 486 486 503 500) * RTP error log - log timestamps (and possibly other data) for each incoming/outgoing RTP packet. This is actually a reasonably chunky piece of work: SIPp doesn’t do any RTP parsing or understand RTP packets at the moment, it just sends them straight through. * Codebase cleanup - the SIPp codebase could be made a little more consistent and friendly to new developers (e.g. better commenting, better documentation of the program flow and structure, more consistency in indentation and whether function names use underscores or camelCase), so I intend to spend some time on this. There's also a possibility that one or two companies will contribute back some in-house changes to SIPp which they've made in the last couple of years, but I'm not yet sure what those changes consist of. I made several commits yesterday, as the start of v3.4 work: these fixed a bug where SIPp used 100% CPU when at its call rate limit, and cleaned up the codebase in various ways (compiling without GCC errors, fixing cppcheck warnings, and formatting). I've broken them down into multiple small commits, so if anyone wants to review them, it shouldn't be an overwhelming job. Best, Rob -- The Go Parallel Website, sponsored by Intel - in partnership with Geeknet, is your hub for all things parallel software development, from weekly thought leadership blogs to news, videos, case studies, tutorials, tech docs, whitepapers, evaluation guides, and opinion stories. Check out the most recent posts - join the conversation now. http://goparallel.sourceforge.net/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- The Go Parallel Website, sponsored by Intel - in partnership with Geeknet, is your hub for all things parallel software development, from weekly thought leadership blogs to news, videos, case studies, tutorials, tech docs, whitepapers, evaluation guides, and opinion stories. Check out the most recent posts - join the conversation now. http://goparallel.sourceforge.net/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp 3.3
Hi Rob, fine job - from my point of view, 3.3b2 works fine. I use it on a Raspi (compiled under Raspbian for ARM) and Ubuntu for continuous voice quality testing and I did not notice any problem. Plans and wishers for 3.4 release? Well, it would be phantastic, if sipp can create a log file containing the arrival times of the media-streams as a csv. E.g., Payloadtype;SSRC;SeqNr;RTPTimestamp;Sourceport;Dstport This could be nice for analysing RTP jitter and RTP packet loss. If you/someone ;) also implements a identical log for the outgoing media streams, the RTP delta could also be calculated. Currently, I solve this task by tracing the media using tcpdump, but in low-performance systems, a second running process (tcpdump) decreases the performance massively. BR Michael Am 05.02.2013 22:52, schrieb Rob Day: Hi all, SIPp 3.3-beta2 has been available for a while now and has a total of around 400 downloads according to Sourceforge, and nobody's reported significant problems either to the list or the bug tracker. As such, I'm going to declare that as the released version - I'll update the version info, and upload renamed tarballs and RPMs, in the next day or two. I'm currently organising my thoughts on what I plan to do for the SIPp 3.4 release, and I'll update the lists as soon as I have something to say on that. In the meantime, if you have some input on that feature list - perhaps a bug you haven't reported yet, or a new feature you'd like, or some code you or your company would like to contribute back to the project - do let me know. Naturally, I'll still accept bug reports and patches throughout the release, but the earlier these things are discussed, the better, since it means there's less risk of contributors duplicating effort. Best, Rob -- Free Next-Gen Firewall Hardware Offer Buy your Sophos next-gen firewall before the end March 2013 and get the hardware for free! Learn more. http://p.sf.net/sfu/sophos-d2d-feb ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Free Next-Gen Firewall Hardware Offer Buy your Sophos next-gen firewall before the end March 2013 and get the hardware for free! Learn more. http://p.sf.net/sfu/sophos-d2d-feb ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp 3.3
Am 05.02.2013 23:22, schrieb Fallon, Josh: With in regards to the media-stream CSV idea, great idea - Thx :) I've been thinking about SIPp on Raspberry Pi as a low cost voice quality probe for a few months now. I wasn't sure how the minimal resources available would handle something like TCPdump while also playing trying to play media, so thanks for sharing that result :) Well, my learnings were: * SIPp (w media) and tcpdump does not produce deterministic results - the system itself creates jitter * Even SIPp (w/o media and w/o tcpdump) has problems when creating 401-authentication responses - here we monitor surprisingly high processing latencies. The funny thing is: the subsequent 407 challenge (in a new SIPp instance, to an INVITE request) is handled deterministic without any spikes. br Michael Josh -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz] Sent: Wednesday, 6 February 2013 9:06 AM To: sipp-users Cc: sipp-devel Subject: Re: [Sipp-users] SIPp 3.3 Hi Rob, fine job - from my point of view, 3.3b2 works fine. I use it on a Raspi (compiled under Raspbian for ARM) and Ubuntu for continuous voice quality testing and I did not notice any problem. Plans and wishers for 3.4 release? Well, it would be phantastic, if sipp can create a log file containing the arrival times of the media-streams as a csv. E.g., Payloadtype;SSRC;SeqNr;RTPTimestamp;Sourceport;Dstport This could be nice for analysing RTP jitter and RTP packet loss. If you/someone ;) also implements a identical log for the outgoing media streams, the RTP delta could also be calculated. Currently, I solve this task by tracing the media using tcpdump, but in low-performance systems, a second running process (tcpdump) decreases the performance massively. BR Michael Am 05.02.2013 22:52, schrieb Rob Day: Hi all, SIPp 3.3-beta2 has been available for a while now and has a total of around 400 downloads according to Sourceforge, and nobody's reported significant problems either to the list or the bug tracker. As such, I'm going to declare that as the released version - I'll update the version info, and upload renamed tarballs and RPMs, in the next day or two. I'm currently organising my thoughts on what I plan to do for the SIPp 3.4 release, and I'll update the lists as soon as I have something to say on that. In the meantime, if you have some input on that feature list - perhaps a bug you haven't reported yet, or a new feature you'd like, or some code you or your company would like to contribute back to the project - do let me know. Naturally, I'll still accept bug reports and patches throughout the release, but the earlier these things are discussed, the better, since it means there's less risk of contributors duplicating effort. Best, Rob -- Free Next-Gen Firewall Hardware Offer Buy your Sophos next-gen firewall before the end March 2013 and get the hardware for free! Learn more. http://p.sf.net/sfu/sophos-d2d-feb ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Free Next-Gen Firewall Hardware Offer Buy your Sophos next-gen firewall before the end March 2013 and get the hardware for free! Learn more. http://p.sf.net/sfu/sophos-d2d-feb ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Free Next-Gen Firewall Hardware Offer Buy your Sophos next-gen firewall before the end March 2013 and get the hardware for free! Learn more. http://p.sf.net/sfu/sophos-d2d-feb ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp 3.3 beta 2
Hi, First of all: thanks for reanimating the SIPp-project and the planned reactivating of the sipp-wiki :) I successfully built sipp 3.3b2 on a raspberry with rasbian for my continuous VoIP-quality testing project. I built it with make pcap_ossl and make all - except a load of the common warning: deprecated conversion from string constant to char* [-Wwrite-strings] warnings, everything worked fine. br Michael Am 13.01.2013 22:08, schrieb Rob Day: Hi all, I've made a few further SVN commits to SIPp, as a result of building it on multiple platforms. Currently I have: * Built on Fedora Linux with PCAP, SCTP and OpenSSL support * Built on FreeBSD with SCTP support * Built on PureDarwin 1.3 with no additional features * Built on Cygwin with PCAP and OpenSSL support I've released this version as sipp3.3-beta2 and uploaded source packages and Centos 6/Fedora 17 RPMs with PCAP and SCTP support. (As a reminder - unlike previous SIPp versions, SIPp 3.3 can handle authentication without needing OpenSSL support.) You can find these files at https://sourceforge.net/projects/sipp/files/sipp/3.3/. I tried for a couple of hours to create a Windows installer for SIPp, but got an Error opening terminal: cygwin error which I wasn't able to fix in the time I had. For now, the best solution for anyone wanting to run SIPp 3.3 on Windows is to install Cygwin and compile and run the code in that environment. Please let me know if you try this beta and you have any feedback (positive or negative - even if you find no bugs, a couple of users saying it works for me will be heartening). Best, Rob -- Master Visual Studio, SharePoint, SQL, ASP.NET, C# 2012, HTML5, CSS, MVC, Windows 8 Apps, JavaScript and much more. Keep your skills current with LearnDevNow - 3,200 step-by-step video tutorials by Microsoft MVPs and experts. ON SALE this month only -- learn more at: http://p.sf.net/sfu/learnmore_123012 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Master Visual Studio, SharePoint, SQL, ASP.NET, C# 2012, HTML5, CSS, MVC, Windows 8 Apps, JavaScript and much more. Keep your skills current with LearnDevNow - 3,200 step-by-step video tutorials by Microsoft MVPs and experts. SALE $99.99 this month only -- learn more at: http://p.sf.net/sfu/learnmore_122412 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Loadtesting Asterisk with Sipp through Kamailio
Hi, this won't work AFAIK - sipp sends requests only to one remote socket during a call. br Michael Am 07.09.2012 08:06, schrieb Grant Bagdasarian: Hello, I’m trying to perform a loadtest on our Asterisk machines using Sipp, but there is a SIP Proxy(Kamailio) in between. Kamailio acts as a loadbalancer for our Asterisk machines, so it only remains in the dialog during the initial INVITE, TRYING and 200 OK. The ACK should be sent by SIPp directly to the Asterisk machines. This is where I’m having problems. I can’t get SIPp to send the ACK directly to the Asterisk machines, without enabling record-route. When I enable record-route Kamailio stays in between, but this is not a representative loadtest for our live platform, since record-route is disabled on live. The contact header is set properly, but SIPp refuses to send the ACK directly. Is this a limitation of SIPp when used in this kind of setup, or am I doing something wrong? I have already loadtested Asterisk without Kamailio in between, that went fine. Regards, Grant -- Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] sipp play pcap file question ,can you help me?
Well, there is one option: you can create a pcap-file containing _both_ audio and video. Then you can use nop action exec play_pcap_audio=audiovideo.pcap/ /action /nop This worked in 90% of all test-pcaps. Sometimes we (our students) noticed, that it does only work if the first packet in the pcap-file is an audio packet, br Michael Am 12/21/2011 02:56 PM, schrieb Greg Thomas: You can only play one pcap file at a time; so you'll need to put a delay between the first and second ones to give the first one time to complete. e.g. nop action exec play_pcap_audio=g711t2.pcap/ /action /nop pause milliseconds=25000/ !-- or however long is required -- nop action exec play_pcap_video=263.cap/ /action /nop Greg 2011/12/21 做些什么好呢 123544...@qq.com: in my uac xml with sipp ,I want to play audio and video pcap file same time ,like following: nop action exec play_pcap_audio=g711t2.pcap/ exec play_pcap_video=263.cap/ /action /nop I found sipp only plays the video file . If i erase the 'exec play_pcap_video=263.cap/ ' ,the audio file can be played successfully,so the two media file are both ok. so ,what can i do to play the two file same time. tks for your answer ,thank you! -- Michael Hirschbichler, Mag. Dipl.-Ing. Institute of Telecommunications Vienna University of Technology A-1040 Wien, Favoritenstr. 9-11/388 Phone: +43 1 58801 38846 -- Write once. Port to many. Get the SDK and tools to simplify cross-platform app development. Create new or port existing apps to sell to consumers worldwide. Explore the Intel AppUpSM program developer opportunity. appdeveloper.intel.com/join http://p.sf.net/sfu/intel-appdev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] unable to parse the msg field
I don't think, that this tag is working. try this: recv request=INVITE optional=true br Michael Am 2011-04-14 09:36, schrieb manasi: recv request=INVITE optional=true -- Benefiting from Server Virtualization: Beyond Initial Workload Consolidation -- Increasing the use of server virtualization is a top priority.Virtualization can reduce costs, simplify management, and improve application availability and disaster protection. Learn more about boosting the value of server virtualization. http://p.sf.net/sfu/vmware-sfdev2dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp Wiki
I have the same problem as well. In beginning of 2011, the wiki was reachable, but outdated and spammed. I contacted Olivier, but also got no response. Got the impression, hp is not more interested in SIPp :( br Michael Am 2011-04-11 09:51, schrieb wondra: Greetings, I am using SIPp in a research project and would be interested in some sample scenarios. However, the SIPp Wiki is out of order (giving http error 500). Is there anyone responsible for the Wiki whom I should contact? I already mailed to Olivier Jacques, but got no response. Thanks, Tomas Vondra -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] successfull calls for server performance
Well, so you just count the sent INVITEs (without retransmissions) and the received ACKs. You can get this information quite easily by using the -trace_shortmsg-function. n(ACK)/n(INVITE w/o retrans) is the successrate ;) br Michael On 2011-03-31 18:10, viswavardhanreddy karna wrote: Hi all, i want to measure depending on the success-rate relevant. i will measure server performance by sending number of calls/second to server to load the server and by that i will be measuring the no:of successful calls On Thu, Mar 31, 2011 at 4:00 PM, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: (Please do not forget to reply to the list too.) Well, what exactly do you want to measure? How do you define performance? * Is the delay relevant (performance indicated by the speed of processing the requests at high load)? * Or is the success-rate relevant (performance indicated by how man calls were set up successfully)? * or is both relevant (a call must handled successfully within a specific time interval)? Depending on your definition, you must decide, where and what you want to observe. br Michael On 2011-03-31 15:26, viswavardhanreddy karna wrote: Hi michael, Thanks again for replying But when you see the only acks ... how many invites you send from client that many acks can be received... so there will be no failure calls every call will be successsful. For calculating the server performance i am loading server with calls/sec of more than 500 or 600 in that case if every call becomes successful how can we know the server performances at acks with regards, viswavardhan On Thu, Mar 31, 2011 at 3:13 PM, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: On 2011-03-31 14:41, viswavardhanreddy karna wrote: I am using SIPp as traffic generator... in order to calculate the number of successful calls from total number of calls which side results should be taken? Well, the question is, how do you define a successful call? And that's the reason why I forwarded you to rfc6076, where some of these metrics are described. E.g., you can calculate the Successful Session Setup - Session Request Delay - the delta between sending the 1st INVITE and receiving the first response initiated by the B-Party (mostly the 180 Ringing). Another metric is the Session Disconnect Delay, where you calculate the delay between sending the BYE and receiving the 200OK Usually, a successful call is initiated, when the ACK arrived at B-Party. So, log with the shortmessages-flag on both A- and B-party and calculate the sent INVITEs-received ACKs ratio br Michael what i mean is user agent client produces some results and user agent server produces some results .. in these 2 results which results should i consider as number of successfull calls? can you please give me an idea regarding this... with regards, viswavardhan On Thu, Mar 31, 2011 at 9:55 AM, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: I propose you take the KPIs from RFC 6076 - Basic Telephony SIP End-to-End Performance Metrics, br Michael On 2011-03-30 23:23, viswavardhanreddy karna wrote: Hi every one, I have a doubt regarding the calculation of server perforrmance. Should we take successfull calls that are obtained from the UAC side ? or successful calls obtained from the UAS SIDE..? for evaluation of server performance... thanking you soo much, with regards
Re: [Sipp-users] successfull calls for server performance
I propose you take the KPIs from RFC 6076 - Basic Telephony SIP End-to-End Performance Metrics, br Michael On 2011-03-30 23:23, viswavardhanreddy karna wrote: Hi every one, I have a doubt regarding the calculation of server perforrmance. Should we take successfull calls that are obtained from the UAC side ? or successful calls obtained from the UAS SIDE..? for evaluation of server performance... thanking you soo much, with regards, viswavardhan -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] successfull calls for server performance
On 2011-03-31 14:41, viswavardhanreddy karna wrote: I am using SIPp as traffic generator... in order to calculate the number of successful calls from total number of calls which side results should be taken? Well, the question is, how do you define a successful call? And that's the reason why I forwarded you to rfc6076, where some of these metrics are described. E.g., you can calculate the Successful Session Setup - Session Request Delay - the delta between sending the 1st INVITE and receiving the first response initiated by the B-Party (mostly the 180 Ringing). Another metric is the Session Disconnect Delay, where you calculate the delay between sending the BYE and receiving the 200OK Usually, a successful call is initiated, when the ACK arrived at B-Party. So, log with the shortmessages-flag on both A- and B-party and calculate the sent INVITEs-received ACKs ratio br Michael what i mean is user agent client produces some results and user agent server produces some results .. in these 2 results which results should i consider as number of successfull calls? can you please give me an idea regarding this... with regards, viswavardhan On Thu, Mar 31, 2011 at 9:55 AM, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: I propose you take the KPIs from RFC 6076 - Basic Telephony SIP End-to-End Performance Metrics, br Michael On 2011-03-30 23:23, viswavardhanreddy karna wrote: Hi every one, I have a doubt regarding the calculation of server perforrmance. Should we take successfull calls that are obtained from the UAC side ? or successful calls obtained from the UAS SIDE..? for evaluation of server performance... thanking you soo much, with regards, viswavardhan -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] DTMF pcap files with payload type=127
Hi, untested: create a SIPp-(UAC)Scenario with a SDP containing only the a common Voice-pt (like 0 or 8) and the pt=127 as you want. Then start Wireshark, run the scenario and call a SIP Hardphone directly. On this hardphone push the buttons in the wished order. In your wireshark-Trace, you should now have the DTMF, hth br Michael On 2011-02-18 06:23, Shah, Nisha N. (Nisha) wrote: Hello, Does anyone know how I can generate DTMF digit pcap files with payload type =127? The digit pcap files that I downloaded from the SIPp website have a payload type=101, and currently I need to test with the 127 payload type. Any help is appreciated. Thanks, Nisha -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE: Pinpoint memory and threading errors before they happen. Find and fix more than 250 security defects in the development cycle. Locate bottlenecks in serial and parallel code that limit performance. http://p.sf.net/sfu/intel-dev2devfeb ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Use of SIPP along with kamailio - REGISTER followed by INVITE not working
Two questions: * the 36000 expiration - is it acknowledged by the Registrar (See Contact-Header in the 200OK response)? * Are you behind a NAT? BR Michael On 2011-01-04 11:24, Stephen McVarnock wrote: Hi, I have got the second scenario here to work i.e. REGISTER xml ran, kill sipp, run sipp with INVITE xml. There seems to be timing issue associated with this though - if I leave to long a delay between killing REGISTER xml and running INVITE xml then the INVITE will not be received by the sipp script. The Expires header for the REGISTER is set to 36000, so that is not the issue. Maybe this is a kamalio problem - anyway, for my purposes (as long as I am quick enough!), this works. Regards, Steve. On 2011-01-04 08:59, Stephen McVarnock wrote: http://sipp.sourceforge.net/wiki/index.php/Patches#Pre.2FPost_scenarios Does anyone know what the current state of play is for this proposed patch or if there is another way to get around this issue? Well, I developed this extension from june until december 2006 but we never managed to merge this branch into the main-tree. 2) I tried to REGISTER the SIPP endpoint in a single xml scenario file with kamailio. This works as per usual. I then killed the SIPP instance and ran a new SIPP script listening on the same port before trying to send the INVITE to it. I expected this to work as the SIPP scenarios (both sending REGISTER and expecting INVITE) listened on the same port. However, the INVITE was not received by the SIPP endpoint. Can anyone think of a reason for this? Well, this is a standard task of sipp and usually works without any limitations. Can you send both scnearios (Register and UAS) as well as the used command line parameters? br Michael -- Learn how Oracle Real Application Clusters (RAC) One Node allows customers to consolidate database storage, standardize their database environment, and, should the need arise, upgrade to a full multi-node Oracle RAC database without downtime or disruption http://p.sf.net/sfu/oracle-sfdevnl ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Subscribe/Notify Call Flow
please do tell. Cheers, Rawat On Thu, Apr 22, 2010 at 21:40, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: Can you send us the xml-scenario and the -trace_msg - Logfile? br Michael On 2010-04-20 08:20, Himanshu Rawat wrote: Hi Michael, I made sample Subscribe/Notify XML scenario and getting below error in sipp error log files. Error log files shows me 200 response received but with below error. * Discarding message which can't be mapped to a known SIPp call:* Scenario: SUBSCRIBE -- 200 -- NOTIFY -- 200 -- In the voice mail server I can see SUBSCRIBE getting processed and 200 response is getting sent. Also Call-id, from tag and CSeq values are same. Any idea why this error is coming ??? Cheers, Rawat On Mon, Apr 19, 2010 at 12:28, Himanshu Rawat himanshu.ra...@gmail.com mailto:himanshu.ra...@gmail.com mailto:himanshu.ra...@gmail.com mailto:himanshu.ra...@gmail.com wrote: Thanks Michale. Tried and worked though with some errors. Will get back if not resolved + more Questions. Cheers, Rawat On Fri, Apr 16, 2010 at 17:46, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: Well, this is not tricky, just send SUBSCRIBE... /send recv response=200/recv recv request=NOTIFY/recv send 200 OK ... /send For the SIP-messages, just start the SIP-client of your choice and record the messages with wireshark br Michael On 2010-04-16 13:55, Himanshu Rawat wrote: Hi Guys, Can anyone tell me on how to write a simple customized Subscribe/Notify Call scenario. Subscribe needs to be send to voice mail server ( which i already have). -- SUBSCRIBE ( to my voice mail server which will process it ) -- 200 [ 50ms] Pause -- NOTIFY ( Will come from my voice mail server ) -- 200 Any link/docs will help :) Cheers, Rawat -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Subscribe/Notify Call Flow
Can you send us the xml-scenario and the -trace_msg - Logfile? br Michael On 2010-04-20 08:20, Himanshu Rawat wrote: Hi Michael, I made sample Subscribe/Notify XML scenario and getting below error in sipp error log files. Error log files shows me 200 response received but with below error. * Discarding message which can't be mapped to a known SIPp call:* Scenario: SUBSCRIBE -- 200 -- NOTIFY -- 200 -- In the voice mail server I can see SUBSCRIBE getting processed and 200 response is getting sent. Also Call-id, from tag and CSeq values are same. Any idea why this error is coming ??? Cheers, Rawat On Mon, Apr 19, 2010 at 12:28, Himanshu Rawat himanshu.ra...@gmail.com mailto:himanshu.ra...@gmail.com wrote: Thanks Michale. Tried and worked though with some errors. Will get back if not resolved + more Questions. Cheers, Rawat On Fri, Apr 16, 2010 at 17:46, Michael Hirschbichler s...@hirschbichler.biz mailto:s...@hirschbichler.biz wrote: Well, this is not tricky, just send SUBSCRIBE... /send recv response=200/recv recv request=NOTIFY/recv send 200 OK ... /send For the SIP-messages, just start the SIP-client of your choice and record the messages with wireshark br Michael On 2010-04-16 13:55, Himanshu Rawat wrote: Hi Guys, Can anyone tell me on how to write a simple customized Subscribe/Notify Call scenario. Subscribe needs to be send to voice mail server ( which i already have). -- SUBSCRIBE ( to my voice mail server which will process it ) -- 200 [ 50ms] Pause -- NOTIFY ( Will come from my voice mail server ) -- 200 Any link/docs will help :) Cheers, Rawat -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net mailto:Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Subscribe/Notify Call Flow
Well, this is not tricky, just send SUBSCRIBE... /send recv response=200/recv recv request=NOTIFY/recv send 200 OK ... /send For the SIP-messages, just start the SIP-client of your choice and record the messages with wireshark br Michael On 2010-04-16 13:55, Himanshu Rawat wrote: Hi Guys, Can anyone tell me on how to write a simple customized Subscribe/Notify Call scenario. Subscribe needs to be send to voice mail server ( which i already have). -- SUBSCRIBE ( to my voice mail server which will process it ) -- 200 [ 50ms] Pause -- NOTIFY ( Will come from my voice mail server ) -- 200 Any link/docs will help :) Cheers, Rawat -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] crazy problem on simple call scenario
This scenario as described below won't work. If I understood the description correctly, the signalling-flow is UA Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- --INVITE- In sipp, the mapping of a message (request/reply) is done by parsing for the SIP Call-ID - if a message is incoming with another call-id than the call-id in the originating request, the message is dropped as an unexpected message. In general, one sipp instance is not able to act as a UAC (for the registration process) and as an UAS (for the incomming invite request) at the same time. You have to split up the functionality to two sequenced sipp-instances: UA_C_ Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- and after that UA_S_ Proxy --INVITE- ---180 ---200 hth and br Michael On 2010-04-09 17:12, Ruhi Aslan wrote: *De :* Ruhi Aslan *Envoyé :* vendredi, 9. avril 2010 16:56 *À :* 'sipp-users-requ...@lists.sourceforge.net' *Objet :* help Hi all, Sipp is a great tool and I currently pull my hair out... I have some trouble with a very simple scenario. I even can't make a call to sipp registered phone. I first registered my phone : sipp -sf callee_hangup_process_test.xml -inf csv/register_client.csv asterisk.ch -trace_err -r1 -m 1 ## register my sipp phone to get calls send ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputerip mailto:1...@mycomputerip CSeq: 1 REGISTER Contact: * Max-Forwards: 5 Expires: 0 User-Agent: SIPp/Linux Content-Length: 0 ]] /send recv response=404 optional=true next=1 /recv recv response=401 auth=true /recv *** Register Process *** send retrans=500 ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputerip mailto:1...@mycomputerip CSeq: 1 REGISTER Contact: * [AUTHENTICATION LINE] Max-Forwards: 5 Expires: 0 User-Agent: SIPp/Linux Content-Length: 0 ]] /send recv response=200 /recv ### phone registered, sip show peer 44 tell me it's OK and reachable on mycomputerIP Then I ask to it to wait until an INVITE comes : recv request=INVITE crlf=true /recv In another window, I make a call with another phone number 43 ( correct scenarios and successfully tested ) sipp -sf callee_hangup.xml -inf csv/caller.cvs asterisk.ch -trace_err -r 1 -m 1 BUT, callee_hangup_process_test.xml doesn't get the INVITE from callee_hangup.xml scenario. The crazy thing is that wireshark says that it sends the expected INVITE to callee_hangup_process_test.xml ( on the right computer, on the right port ). But on my previous INVITE recv request, the count persist on 0 ! Here the INVITE sended to mycomputerIP ( supposed to make the INVITE recv reauest count up to 1 ) INVITE sip:4...@mycomputerip:5060 SIP/2.0 Record-Route: sip:sipproxy;lr=on;ftag=ftag;vsf=some...;did=... Via: SIP/2.0/UDP sipproxy;branch=z9hG4bK-ID2 Via: SIP/2.0/UDP asterisk.ch:5060;received=asterisk.ch;branch=z9hG4b-ID;rport=5060 From: 43 sip:4...@voip.vtx.ch;tag=as1cf8af76 To: sip:4...@mycomputerip:5060 Contact: sip:4...@_asterisk.ch_ Call-ID: call...@asterisk.ch mailto:call...@asterisk.ch CSeq: 102 INVITE User-Agent: voipua Max-Forwards: 69 Date: Fri, 09 Apr 2010 13:54:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 242 P-hint: outbound v=0 o=root 26199 26199 IN IP4 _asterisk.ch_ s=session c=IN IP4 _asterisk.ch_ t=0 0 m=audio 18150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - more info : I already use -aa option for OPTIONS NOTIFY request, and on the second OPTIONS, sipp crash on seg fault :-\ So where is my mistake ? Ruhi ASLAN Stagiaire ST40 - NOC/Operation -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] about sipp capacity
Hi, 408cps (SIP only, no media) are definetly no problem for a standard UAS-scenario on a current hardware-configuration. Furthermore, a UAS does not generate error-responses automatically by itself (except a BYE-response if the -nd - flag is not set), br Michael WANG Jin jia Jw wrote: To Whom It May Concern: I am using the SIPp v3.1-TLS to simulate the IMS 2 IMS call load. I use serverA and serverB as UAC and serverC as UAS. All UAC generated calls are terminated at the UAS. The rate of call is up to 408 cps in total and the call hold time is 145 seconds. And I found the CPU usage of the serverC increased to above 90% and call began to fail with 408, 480, 503 error code in the UAC part. It seemed sipp was in overload status under such load. My questions are: What is the sipp peak capacity and what will sipp react when it is overload? Thanks for your help! Jinjia -- The Planet: dedicated and managed hosting, cloud storage, colocation Stay online with enterprise data centers and the best network in the business Choose flexible plans and management services without long-term contracts Personal 24x7 support from experience hosting pros just a phone call away. http://p.sf.net/sfu/theplanet-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- The Planet: dedicated and managed hosting, cloud storage, colocation Stay online with enterprise data centers and the best network in the business Choose flexible plans and management services without long-term contracts Personal 24x7 support from experience hosting pros just a phone call away. http://p.sf.net/sfu/theplanet-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp with Open IMS Core, OK message does not arrive from IMS core
Hi, Your Call-ID is 1-17...@127.0.0.1, where a call-id in sipp in general is composed of 'number of call'-'processid'@'local_ip'. So, IMHO, sipp takes the wrong, the loopback, IP-address for the variable [local_ip]. Try sipp again and use the -i-parameter to explicitely define the IP adress which should be used. br Michael PS: as far as I know, sipp uses the same function to detect the primary IP-address like the linux-command hostname -i. So, when hostname -i results in 127.0.0.1, you will get the mentioned problem- mustafa rifaee wrote: Hello all; I am using SIPp 3.1 with Open IMS Core, but when i send registration the registration done successfully but the IMS core does not send OK message back to SIPp. *Aborting call on unexpected message for Call-ID '1-17420 at 127.0.0.1 https://lists.berlios.de/mailman/listinfo/openimscore-users': while expecting '200' response, received 'SIP/2.0 401 Unauthorized - Challenging the UE* Please Help me. Best Regards Rifaee --. registration scenario:- ?xml version=1.0 encoding=ISO-8859-1 ? !DOCTYPE scenario SYSTEM sipp.dtd scenario name=reg_alice send retrans=500 ![CDATA[ REGISTER sip:open-ims.test SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] Route: sip:pcscf.open-ims.test:4060; lr Max-Forwards: 70 From: alice sip:al...@open-ims.test:4060 To: alice sip:al...@open-ims.test:4060 P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E Call-ID: [call_id] Contact: sip:al...@[local_ip]:[local_port];transport=[transport] Content-Length: 0 Supported: path Expires: 300 CSeq: 1 REGISTER User-Agent: Sipp v1.1-TLS, version 20061124 ]] /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:open-ims.test SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] Route: sip:pcscf.open-ims.test:4060;lr Max-Forwards: 70 From: alice sip:al...@open-ims.test To: alice sip:al...@open-ims.test P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E Call-ID:[call_id] CSeq: 2 REGISTER Contact: sip:al...@[local_ip]:[local_port] Expires: 300 Content-Length: 0 [authentication username=al...@open-ims.test password=alice] Supported: path User-Agent: Sipp v1.1-TLS, version 20061124 ]] /send recv response=200 /recv /scenario -- Throughout its 18-year history, RSA Conference consistently attracts the world's best and brightest in the field, creating opportunities for Conference attendees to learn about information security's most important issues through interactions with peers, luminaries and emerging and established companies. http://p.sf.net/sfu/rsaconf-dev2dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Throughout its 18-year history, RSA Conference consistently attracts the world's best and brightest in the field, creating opportunities for Conference attendees to learn about information security's most important issues through interactions with peers, luminaries and emerging and established companies. http://p.sf.net/sfu/rsaconf-dev2dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] successful dead call issue
Cool, SIPp, IMS and IPv6 ;) Hmm, my guess is, that the ACK is not recognized correctly by the IMS-core-logic and the dead-call messages are retransmissions. As first debugging step, I propose to add a pause of a few seconds at the end of the UAC-scenario. Then, you will see, if there are any retransmissions arriving after sending the ACK-request. br Michael WANG Jin jia Jw wrote: To Whom It May Concern: I am running SIPp v3.1-TLS. Following figure is my scenarios (call forwarding). We have IMS core and a sip application deployed. As you see, there are many successful dead call in UAC part. Is that normal? How to avoid such error? Would you please help to analyze this? I am attaching the UAC and UAS scripts and scenarios for you. Thanks a lot for your help! Jinjia -- Return on Information: Google Enterprise Search pays you back Get the facts. http://p.sf.net/sfu/google-dev2dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Return on Information: Google Enterprise Search pays you back Get the facts. http://p.sf.net/sfu/google-dev2dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp can't send RTP pckts to AST
IIRC there was a substr-Bug in earlier sipp-versions. Try updating to the newest version, br Michael srt_liyq schrieb: 1. SIPp sends Invite（with sdp） 2. Receive the Response Message 200 OK from AST， including Media Description, name and address : audio 10178 RTP/AVP 8 101. 3. but SIPp sends UDP data package to the No. 1017 port, that is, gets 4 leading digits of the negotiated port? -- Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] 407 for INVITE
OK, imho: obviously, the call setup succeeded after beeing challenged by your x-cscf - this means, the challenge in the 407 - Response was interpreted correctly and used for the 2nd INVITE-request by the sipp uac. The x-cscf accepted the credentials and forwards the INVITE-request to the sipp-uas. So, from cscf-side, there is no reason to send a 407 to the second invite request. So, there might be two explanations: either, you have a completely non-3261 conform core, which does something completely wrong, or the 407ers are retransmissions of the first one according to the 1st INVITE. How to find this out? Well, the retransmitted 407 are absolutely identical to the the first 407 and arrive 0.5, 1.5, 3.5 and 7.5 seconds after the first 407, br Michael mwilliam prusty schrieb: Hi Michael My S-cscf is configured in such a way that , until it gets the credentials for the INVITE request , it will not pass the INVITE request to other endpoint, Here After getting the Second INVITE(cseq=2)with authorizatoion, header field, it is forwarding that INVITTE request to other end point. In SIPP server side i am able to cptutre the messages after the SIPP server getting the INVITE, it is sending 180, 200, ACK. RTP packets also being exchange. I am able to termicate the call also using the BYE. reueest. After the call terminated CSCF is keep on sending the 407 response . One more thing You have mentioned there might be wrong with the ACK for the 407 resposne. How i can check that Regards William On Thu, 29 Oct 2009 14:51:36 +0530 wrote Are you sure, the 407 is for the second INVITE with the credentials? Are you getting a 100/180/183/200 for the second INVITE-request? My guess is, that your ACK for the 407 is not correct and the subsequent 407ers are retransmissions. br Michael PS: please no x-posting. mwilliam prusty wrote: Hi All I am using SIPP for IMS. Here i am facing issue which is given as follows. 1. Client sending INVITE ,getting 407. 2. After that Client sending INVITE(cseq=2) With proper credentilas 3. After that Call is estyablished between the Endpoints. 4. After Client is sending BYE, the SUTis keep on 407 for the second INVITE(cse=2). But using Xlite hard phones its working fine . the i.e P-csf is not keep on sending 407 for INVITE(cseq=2) Any guides. william -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] 407 for INVITE
Are you sure, the 407 is for the second INVITE with the credentials? Are you getting a 100/180/183/200 for the second INVITE-request? My guess is, that your ACK for the 407 is not correct and the subsequent 407ers are retransmissions. br Michael PS: please no x-posting. mwilliam prusty wrote: Hi All I am using SIPP for IMS. Here i am facing issue which is given as follows. 1. Client sending INVITE , getting 407. 2. After that Client sending INVITE(cseq=2) With proper credentilas 3. After that Call is estyablished between the Endpoints. 4. After Client is sending BYE, the SUTis keep on 407 for the second INVITE(cse=2). But using Xlite hard phones its working fine . the i.e P-csf is not keep on sending 407 for INVITE(cseq=2) Any guides. william On Thu, 29 Oct 2009 00:07:27 +0530 wrote Hi, I found this thread because I have got the same problem. Please, Mosbah, If you resolved the registration process, tell us how did you do it, because it could be helpful for other people. I attached my sipp script and the wireshark information of my UE.Thanks in advace,On Mon, Jun 22, 2009 at 10:45 AM, Kirwan, David (David) wrote: From the RFC 3261 SIP: Session Initiation Protocol 21.4.2 401 Unauthorized The request requires user authentication. This response is issued by UASs and registrars, while 407 (Proxy Authentication Required) is used by proxy servers. The request you are sending does not contain the correct authentication information, please attached your scenario file. From: mosbah agil [mailto:mmaas...@yahoo.com] Sent: 20 June 2009 12:26To: IMS; Franz edler; Richard GoodCc: SIPp mailing listSubject: [Sipp-users] IMS-SIPp registration error Hi all,we try to register on IMS using SIPp but this error appear in the SIPp terminal, if any one know what we should do to overcome this error, please help me. best regards,Mosbah.2009-06-20 12:21:34: Aborting call on unexpected message for Call-ID '1-7...@192.168.100.22': while expecting '200' response, received 'SIP/2.0 401 Unauthorized - Challenging the UEVia: SIP/2.0/UDP 192.168.100.22:5060;rport=5060;branch=z9hG4bK-7118-1-2From: bob ;tag=1To: bob ;tag=d06116319426125fe9069e4ac62d5a83-d94eCall-ID: reg///1-7...@192.168.100.22cseq: 2 REGISTERPath: Service-Route: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFOServer: Sip EXpress router (2.1.0-dev1 OpenIMSCore (i386/linux))Content-Length: 0Warning: 392 192.168.100.14:6060 Noisy feedback tells: pid=6413 req_src_ip=192.168.100.14 req_src_port=5060 in_uri=sip:scscf.hii2-ims.test:6060 out_uri=sip:scscf.hii2-ims.test:6060 via_cnt==3WWW-Authenticate: Digest realm=hii2-ims.test, nonce=8NejVST30rdbKHMyLjUYtEPLEkZUfAAA/lcHsC/j9CY=, qop=auth, algorithm=AKAv1-MD5 -- Are you an open source citizen? Join us for the Open Source Bridge conference! Portland, OR, June 17-19. Two days of sessions, one day of unconference: $250. Need another reason to go? 24-hour hacker lounge. Register today! http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Vanessa Tejada -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline@middle? -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only
Re: [Sipp-users] UAS not quitting after reaching call-limit
Hi all, am I the only one having this problem? Does anyone know how to solve this issue? br Michael (Xposted and follow-up set to devel-list) Michael Hirschbichler wrote: Hi all, I am currently using sipp30 (SIPp v3.1-TLS-PCAP, version svn522M) and when running as UAS, it does not automatically quit after receiving -m-Requests. Another sipp-Version (SIPp v2.0.1-TLS-PCAP, version 20070516) is quitting correctly. So, well, the question: is it a bug, or a feature? How can I change to the usual behaviour? br Michael -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] UAS not quitting after reaching call-limit
Hi all, I am currently using sipp30 (SIPp v3.1-TLS-PCAP, version svn522M) and when running as UAS, it does not automatically quit after receiving -m-Requests. Another sipp-Version (SIPp v2.0.1-TLS-PCAP, version 20070516) is quitting correctly. So, well, the question: is it a bug, or a feature? How can I change to the usual behaviour? br Michael -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Understanding RTP in SIPP
Hi all, I am using a pcap sound sample to be replayed with sipp with nop action exec play_pcap_audio=./mediastream.pcap/ /action /nop This media-stream in the pcap-file has some jitter and various inter-packet-delay. Am I correct in the assumption, that this jitter and delay will not have any effect on the pcap-replay functionality of sipp? Or will the pcap-file be replayed as is with the same jitter and delay as in the original trace? Thanks in advance and best regards Michael -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Understanding RTP in SIPP
Hi Bradley, inter packet delay is the delta between arriving of each of the RTP-packets. For example, each rtp packet is sent 20ms after the packet before. On B-Party side, the time between the received packets may vary due to network delay - this additional delay is responsible for the jitter. My question was: Will on a replayed pcap file the delta between each of the rtp packets be automatically normalized to these 20ms or will it be sent with the delta as reveiced and recorded? Or in other words: is the RTP-timestamp or the time-of-arrival used as time-reference for replaying the stream? br Michael Bradley, Todd wrote: Yes, it'll still have the same jitter. In other words, if you captured the packets out of order, they'll still be out of order when you play them back. But what does delay mean in this example? The gap in time between when the packet was sent and when it arrives? That's going to be days, weeks, or months in this case. Cheers, Todd. -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz] Sent: Tue 10/20/2009 3:56 AM To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] Understanding RTP in SIPP Hi all, I am using a pcap sound sample to be replayed with sipp with nop action exec play_pcap_audio=./mediastream.pcap/ /action /nop This media-stream in the pcap-file has some jitter and various inter-packet-delay. Am I correct in the assumption, that this jitter and delay will not have any effect on the pcap-replay functionality of sipp? Or will the pcap-file be replayed as is with the same jitter and delay as in the original trace? Thanks in advance and best regards Michael -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Starting sipp paused?
Hi all, I want to start multiple sipp instances in parallel. To synchronise them, I am planning to use the remote control UDP-socket. So, I want to start each instance one after another in the paused state and then I want to send each of them the p-letter to start the traffic as synchronised as possible. Is there a way to do this? br Michael -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Starting sipp paused?
Hi David, thanks, I know the IMS-bench (it came too late for my needs a few years ago ;-) ). For my current test, it is quite an overkill - is there a way to do this job also with the plain sipp? br Michael Verbeiren, David wrote: You may know this already but IMS Bench SIPp has this behaviour built in, with its manager process controlling the SIPp instances. See http://sipp.sourceforge.net/ims_bench Regards, -David -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz] Sent: mercredi 16 septembre 2009 14:21 To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] Starting sipp paused? Hi all, I want to start multiple sipp instances in parallel. To synchronise them, I am planning to use the remote control UDP-socket. So, I want to start each instance one after another in the paused state and then I want to send each of them the p-letter to start the traffic as synchronised as possible. Is there a way to do this? br Michael -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - Intel Corporation NV/SA Rond point Schuman 6, B-1040 Brussels RPM (Bruxelles) 0415.497.718. Citibank, Brussels, account 570/1031255/09 This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies. -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] sendto failed with error: Address family not supported by protocol.
catalina oancea wrote: I also tried with snapshot http://sipp.sourceforge.net/snapshots/sipp.2009-01-21.tar.gz. The same problem occurs. The sipp command is: /usr/local/sipp//sipp -sf scen.xml -t un -r 20 -l 200 -aa -i 192.168.13.13 -m 1000 -inf cases.csv -trace_rtt -trace_screen -trace_stat -trace_msg -trace_logs -fd 4 -max_retrans 15 192.168.12.12 -trace_err Try the remote IP as last parameter: /usr/local/sipp//sipp -sf scen.xml -t un ... -trace_err 192.168.12.12 BR Mike Anybody? -- This SF.net email is sponsored by: High Quality Requirements in a Collaborative Environment. Download a free trial of Rational Requirements Composer Now! http://p.sf.net/sfu/www-ibm-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- This SF.net email is sponsored by: High Quality Requirements in a Collaborative Environment. Download a free trial of Rational Requirements Composer Now! http://p.sf.net/sfu/www-ibm-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Bug in -bind_local parsing?
Hi all, I noticed a strance behaviour when passing an -bind_local - Argument: Following the online-help: ./sipp -h the remotehost must be added as first argument: sipp remote_host[:remote_port] [options] entering ./sipp 2.2.2.2 -sn uac -bind_local 1.2.3.4 results in --- Scenario Screen [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 5060 1.70 s 17 1.2.3.4:5060(UDP) which is definitively the wrong remotehost if I add the remotehost as the last argument, ./sipp -sn uac -bind_local 1.2.3.4 2.2.2.2 then the behavior is correct: --- Scenario Screen [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 5060 0.89 s8 2.2.2.2:5060(UDP) This error affects at least version 3.1 (svn803) and sipp 2.0 (20070426) BR Michael - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] [ sipp-Patches-1823593 ] raw sockets for spoofing source IP address/port
Hi all! I wanted to use this patch, but am I correct, that it is currently not merged with the main tree? I also tried to patch the diff against trunk-rev. 332 (as used in the diff-file : --- sipp.hpp(revision 332) +++ sipp.hpp(working copy) Surprisingly, also against rev. 332, the patch fails - especially in the sipp.hpp and sipp.cpp are a lot of differences. Can anybody give me a hint to get this thing done? br Michael SourceForge.net schrieb: Patches item #1823593, was opened at 2007-10-31 18:04 Message generated for change (Tracker Item Submitted) made by Item Submitter You can respond by visiting: https://sourceforge.net/tracker/?func=detailatid=637566aid=1823593group_id=104305 Please note that this message will contain a full copy of the comment thread, including the initial issue submission, for this request, not just the latest update. Category: None Group: None Status: Open Resolution: None Priority: 5 Private: No Submitted By: Klaus Darilion (klaus_darilion) Assigned to: Nobody/Anonymous (nobody) Summary: raw sockets for spoofing source IP address/port Initial Comment: Hi! This patch adds support for sending UDP messages with spoofed IP address and UDP port (please review the code): regarding the patch for call.cpp: the first 2 chunks are only indent fixes - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Asterisk and Authoriation
Just increase the CSEQ-Number of the 2nd INVITE (message 5), BR Michael d 82 k schrieb: Hi everybody, I would like to test my asterisk and in order to do this I would like to run sipp on two computers (A and B) and register some users (1001 : 1010 for A and 2001 : 2010 for B) and make users A call user B. I have created a scenario to register user in a sequential way, and it seems to work. I have also used the default scenario UAC and edited like this for users A: SIPp UACRemote |(1) INVITE | |--| |(2) 100 (optional) | |--| |(3) 407| |--| |(4) ACK| |--| |(5) INVITE | |--| |(6) 100 (optional) | |--| |(7) 180 (optional) | |--| |(8) 200| |--| |(9) ACK| |--| | | |(10) PAUSE | | | |(11) BYE | |--| |(12) 200 | |--| send ![CDATA[ INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[field2];branch=[branch] From: [field0] sip:[EMAIL PROTECTED];tag=[call_number] To: [field3] sip:[EMAIL PROTECTED] Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:[field2] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]] /send recv response=100 optional=true /recv recv response=407 auth=true /recv send ![CDATA[ ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[field2];branch=[branch] From: [field0] sip:[EMAIL PROTECTED];tag=[call_number] To: [field3] sip:[EMAIL PROTECTED] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:[EMAIL PROTECTED]:[field2] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]] /send send ![CDATA[ INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[field2];branch=[branch] From: [field0] sip:[EMAIL PROTECTED];tag=[call_number] To: [field3] sip:[EMAIL PROTECTED] Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:[field2] Max-Forwards: 70 [authentication username=601 password=1234] Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]] /send recv response=100 optional=true /recv recv response=180 optional=true /recv [...] and I'm using default UAS for users B. the problem is that everything goes fine untill message 5, the new invite with the authentication response is sent and the server seems to not accept it. It seems it doesn't recognize the message... (Maybe is the call id different between the invite 1 and 5... I can't check it right now...) Any ideas? how can I solve this? Thankyou for your help and time dk Connect and share in new ways with Windows Live. Get it now! http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008 - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIPp discards if it receives extra header in the response
You have to let SIPp generate the Call-ID: --- [call_id] A call_id identifies a call and is generated by SIPp for each new call. In client mode, it is mandatory to use the value generated by SIPp in the Call-ID header. Otherwise, SIPp will not recognise the answer to the message sent as being part of an existing call. --- (http://sipp.sourceforge.net/doc2.0/reference.html#Structure+of+client+%28UAC+like%29+XML+scenarios) BR Michael Santosh Reddy wrote: Hi, I am doing a basic registration scenario where SIPp sends REGISTER and receives 200 OK with some additional headers. I have attached the error and message logs. please look into it. Can SIPp ignore or receive some custom header received in responses -- 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
[Sipp-users] using Retry After:-Header
Hi listmembers! I am working on a load-test for a registrar, but after a very high load, I get a 503 Server too busy with a Retry After: 10-Header. Does anyone in this list ever reused this header by grepping the numeric part out, and using it with a next-field and a Pause-command - is it possible to reuse the value of a variable in the pause-command? Something like this: send ... REGISTER ... /send recv response=200 optional=true next=2 / recv response=503 action !-- Search for retry-time -- ereg regexp=*** search_in=hdr header=Retry After: check_it=true assign_to=2 / /action /recv pause milliseconds=$2/ send ... REGISTER ... /send label id=2/ Thanks in advance Michael PS: also thanks to the users, who posted their SIPp-cps - high-scores! I think, Charles Wright made the race with 10.000 calls per second. -- Michael Hirschbichler, Dipl.-Ing. Institut fuer Breitbandkommunikation Technische Universitaet Wien A-1040 Wien, Favoritenstr. 9-11/388 Tel: +43 1 58801 38846 - 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
[Sipp-users] SIPp Performance - High-Score?
Hi all! Just had a (awkward) discussion with a representative of a commercial SIP-testing - solution about the performance of SIPp. Now I am wondering about Your experiences for the maximum measured cps-rate of a 'usual' INVITE-scenario - just for statistics. I'm not curious about the performance - or the type - of the SUT, but just the current cps - high-score, You ever archived with SIPp. I think, this fact would also be useful for the wiki, thanks in advance and BR Michael -- Michael Hirschbichler, Dipl.-Ing. Institut fuer Breitbandkommunikation Technische Universitaet Wien A-1040 Wien, Favoritenstr. 9-11/388 Tel: +43 1 58801 38846 - 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
[Sipp-users] fieldn in authentication-header
Hi all! I just updated SIPp to the most current version, but as I wanted to run my scenarios, I noticed, that the auth-error is back again: the line [authentication username=[field4] password=[field1]] creates as a result this SIP-Header: sip_test_user_1[authentication username=I300364267P257341267^X326363267364217365267^H5362267 password=] I submitted a patch nearly a half year ago, fixing this bug (changelog.txt: 2007-04-25), but now it seems to be back again. Can anyone confirm this, or is my xml-syntax incompatible with the current version? BR Michael - 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] Question: Which VoIP-testools do You use?
Thanks Klaus! Your page was one of the ressources I already used for my recherche :-) Have You (or someone else) ever used TTC-N3 for SIP-testing? The product of Testing-Technologies (TTWorkbench) seems to be interesting, doesn't it? We're discussing about buying a license for our institute ... BR Michael Hi! I also use openser. Configure your SIP client to use an openser as outbound proxy. Then you can do all the nice message manipulation in openser by using your favorite SIP client. IMO eyebeam is an absolute MUST for SIP testing. Lots of features, multiple lines, very stable and you can configure a lot of things. For debugging: ngrep (ngrep -t -W byline -d any port 5060 or or port 5061 or port 53 or icmp is my favorite) wireshark (especially for RTP analysis) you can also find some tools here: http://www.pernau.at/kd/voip/bookmarks-sip-test.html regards klaus Michael Hirschbichler wrote: Hi all, I am working on an overview about different VoIP/SIP/IMS - related testtools, and I am wondering, which tools(GPL and non-free) are You using. Well, some tools I already found during my recherche: * The great and wonderful SIPp ;-)) * Protos-Test - Suite * Spirent IMS Opticom Pesq * Sipsak * SIP Proxy Any other proposals and experiences? Best regards Michael - 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: Which VoIP-testools do You use?
Thanks Klaus! Your page was one of the ressources I already used for my recherche :-) Have You (or someone else) ever used TTC-N3 for SIP-testing? The product of Testing-Technologies (TTWorkbench) seems to be interesting, doesn't it? We're discussing about buying a license for our institute ... BR Michael Hi! I also use openser. Configure your SIP client to use an openser as outbound proxy. Then you can do all the nice message manipulation in openser by using your favorite SIP client. IMO eyebeam is an absolute MUST for SIP testing. Lots of features, multiple lines, very stable and you can configure a lot of things. For debugging: ngrep (ngrep -t -W byline -d any port 5060 or or port 5061 or port 53 or icmp is my favorite) wireshark (especially for RTP analysis) you can also find some tools here: http://www.pernau.at/kd/voip/bookmarks-sip-test.html regards klaus Michael Hirschbichler wrote: Hi all, I am working on an overview about different VoIP/SIP/IMS - related testtools, and I am wondering, which tools(GPL and non-free) are You using. Well, some tools I already found during my recherche: * The great and wonderful SIPp ;-)) * Protos-Test - Suite * Spirent IMS Opticom Pesq * Sipsak * SIP Proxy Any other proposals and experiences? Best regards Michael - 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
[Sipp-users] [branch]-struggle
Hi Group! Currently I am trying to create an UAC with the following call-flow UAC Proxy INVITE--- --100 Trying --407 Proxy Auth Required ACK-- INVITE (w.auth)-- The problem is the ACK-request: send rtd=true ![CDATA[ ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/[transport] [local_ip];branch=[branch] CSeq: 1 ACK ... ! /send By using the [branch]-field as described in the help, I get as a result a new branch-parameter z9hG4bK-1-4, but I _must_ have the same branch as in the INVITE: RFC3261=== 17.1.1.3 Construction of the ACK Request This section specifies the construction of ACK requests sent within the client transaction. A UAC core that generates an ACK for 2xx MUST instead follow the rules described in Section 13. The ACK request constructed by the client transaction MUST contain values for the Call-ID, From, and Request-URI that are equal to the values of those header fields in the request passed to the transport by the client transaction (call this the original request). The To header field in the ACK MUST equal the To header field in the response being acknowledged, and therefore will usually differ from the To header field in the original request by the addition of the tag parameter. The ACK MUST contain a single Via header field, and this MUST be equal to the top Via header field of the original request. ... == Currently, I am planning to use the regexp to parse this parameter. Does anyone has a better idea? What about extending the [last_...]-feature by creating something like: [last_header_field] or [last_header_index_field], like [last_Via_0_brach]? Best regards Michael -- Michael Hirschbichler, Dipl.-Ing. Institut fuer Breitbandkommunikation Technische Universitaet Wien A-1040 Wien, Favoritenstr. 9-11/388 Tel: +43 1 58801 38846 - 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
[Sipp-users] Listen to more than one UDP-Socket?
Hi! I want to use sipp in the current version to listen to more than one local UDP-Port. Is this possible in some way? Background: I register 1500 different user from the same host, but with different port-numbers in the Contact:- and the Via:-Header field. After registering, I create 1500 calls, one to each of these users. For this reason, I want sipp to listen to port 6050 to 7550 to answer these requests ... Best regards Michael - 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
[Sipp-users] [fieldn] int the [authentication ...]-part
Hi all! I am trying to inject data from a .csv-file into the [authentication ...]-line of a REGISTER-request. Am I correct in the assumption, that this isn't working in the current rc8? Example: The xml-line [authentication username=[field0] password=[field1]] creates: Authorization: Digest username=[field0,realm=provider.net,cnonce=6b8b2567,nc=0001,qop=auth,uri=sip:xxx.xxx.xxx.xxx:5060,nonce=9824f492e7c5e3bd41d7c12372e56e,response=cde60345de05509b99cea94021e562aa73,algorithm=MD5 password=pass] BR Michael - 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
Hi I didn't test your XML-file, but you have at least one bug: The correct syntax is recv response ... /recv send ... #SIP code ... /send Your /recv-tag is located at the end of the xml-snipplet and that's wrong :-) Greets Michael Federico La Volpe wrote: Hi guys, I am new on this group. I am testing the Sipp, but I have a problema, when the Sipproxy sent to the SIpp UAC the 407, it does not response with the new invite includind the Digest Someone knows what is wrong? here i copy the code thank you recv response=407 auth=true send ![CDATA[ INVITE sip:[EMAIL PROTECTED] mailto:sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port] From: 1151681326 sip:[EMAIL PROTECTED]:[local_port];tag=[call_number] To: sip:[EMAIL PROTECTED] mailto:sip:[EMAIL PROTECTED] Call-ID: [call_id] Cseq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:[local_port] [authentication username=1151681326 password=fede2399] Max-Forwards: 70 Content-Type: application/sdp Subject: Performance Test Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=shit t=0 0 c=IN IP[media_ip_type] [media_ip] m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]] /send /recv - 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
