u can try sumthing like this....i guess support for pcap is not avail in windows. on linux its this. run -trace_logs -r 1 -rp 50000 -sn uac_pcap 192.168.75.32:50745 provided u hv configured pcap lib.
On 8/12/08, Srivastava, Anuj Kumar <[EMAIL PROTECTED]> wrote: > Sending your scenario file would help. > > > ________________________________ > From: darshan b n [mailto:[EMAIL PROTECTED] > Sent: Tuesday, August 12, 2008 12:59 PM > To: Srivastava, Anuj Kumar > Cc: [email protected] > Subject: Re: [Sipp-users] Sipp-users Digest, Vol 27, Issue 3 > > That i have done but it is giving following error: > > /ats/tools/sipp-2.0.1.src> ./sipp 10.34.20.78<http://10.34.20.78> -sf > /home//CQ78827_UPDATE_pcap2.xml -m 1 -i 10.128.254.81<http://10.128.254.81> > -p 6060 -mp 10000 > > In pcap pcap/g711a.pcap, npkts 236 > max pkt length 260 > base port 2006 > In pcap pcap/dtmf_2833_1.pcap, npkts 10 > max pkt length 24 > base port 10000 > > > > > On 12/08/2008, Srivastava, Anuj Kumar > <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> wrote: > Sipp should be compiled with pcap option. > there's no specific command for executing pcap scenarios. > > regards > Anuj > > ________________________________ > From: > [EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]> > [mailto:[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>] > On Behalf Of darshan b n > Sent: Tuesday, August 12, 2008 12:09 PM > To: > [email protected]<mailto:[email protected]> > Subject: Re: [Sipp-users] Sipp-users Digest, Vol 27, Issue 3 > > > i want know the what is the command to excecute sipp with PCAP > > Regards > darshan > > > On 08/08/2008, > [EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]> > <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> > wrote: > Send Sipp-users mailing list submissions to > > [email protected]<mailto:[email protected]> > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/sipp-users > or, via email, send a message with subject or body 'help' to > > [EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]> > > You can reach the person managing the list at > > [EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]> > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Sipp-users digest..." > > > Today's Topics: > > 1. Re: SUBSCRIBE scenario (??????? ?.?.) > 2. SIPp problem with pcap ( Jes?s Camacho Rodr?guez ) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 07 Aug 2008 18:35:49 +0400 > From: ??????? ?.?. <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> > Subject: Re: [Sipp-users] SUBSCRIBE scenario > To: > [email protected]<mailto:[email protected]>, > ims.asuser ims.asuser" > <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> > Message-ID: > <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> > Content-Type: text/plain; charset=windows-1251 > > > Hi Khaldon, > > i did perfomance tests of OpenIMSCore using SIPp 2.1 and SIPp IMS Bench. > Normal Registration signaling flow looks like: > >> 1) register -> >> 2) <-401 >> 3) register-> >> 4) <-200 > > it's enough to provide the registration for user. Pay attention to P-Headers > ;) > > In addition you'll need to fill the FoHSS with a number of subscriber > entities, to handle this issue in automatic mode analyse the SQL scripts > used for creation of Alice and Bob. > > Regards, > Ivan Kuzmin. > State University of Telecomunications, St.Petersburg, Russia. > > > -----Original Message----- > From: "ims.asuser ims.asuser" > <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> > To: > [email protected]<mailto:[email protected]> > Date: Thu, 7 Aug 2008 16:16:40 +0200 > Subject: [Sipp-users] SUBSCRIBE scenario > >> >> Hi all, >> >> I would like to test the OpenIMSCore platform (I will perform as much as I >> can). These tests will be then given to the IMS community. >> First of all, did anyone perform OpenIMSCOre tests using SIPp? >> >> I would like to perform first in different stage: >> 1) Registration >> 2) Calls >> 3) Instant Messaging >> >> So far, I'm only on the first step (registering a user). I only managed to >> do a simple registration (register ->, <-401, register->, <-200). But I >> have >> problems when I want to go further. >> In ims, the scenario looks like that: >> 1) register -> >> 2) <-401 >> 3) register-> >> 4) <-200 >> 5) subscribe-> >> 6) <-notify >> 7) 200-> >> >> The IMS Core is behaving really eerily when I send a subscribe message >> then >> the simulation stops >> Is anyone know why is it working like that? >> >> Attached are my script and wireshark trace. >> >> Many thanks, >> Khaldon >> >> ATTACHMENT: text/xml (Registerbob1.xml) >> ATTACHMENT: application/octet-stream (SIPp_SUBSCRIBE) >> >> ------------------------------------------------------------------------- >> 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=100&url=/ >> >> _______________________________________________ >> Sipp-users mailing list >> [email protected]<mailto:[email protected]> >> https://lists.sourceforge.net/lists/listinfo/sipp-users >> >> > > > > > ------------------------------ > > Message: 2 > Date: Thu, 7 Aug 2008 20:33:24 +0200 > From: " Jes?s Camacho Rodr?guez " > <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> > Subject: [Sipp-users] SIPp problem with pcap > To: > [email protected]<mailto:[email protected]> > Message-ID: > > <[EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]>> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all, > > I am doing Asterisks performance tests on a pico-itx box. I compiled SIPp > with pcap play support in a Debian Etch machine so I could send RTP packages > through a recorded pcap file for simulating traffic. I run SIPp in client > mode in one machine and I call to a VoIP phone (actually I wanted to call to > a SIPp in server mode but I am calling to a phone for debugging purposes). > The problem I have is that SIPp in client mode sends the RTP packages to > Asterisk (at least that is what SIPp shows in the "Total RTP pckts sent"), > but if I activate the RTP debug mode in Asterisk, I see that the > communication is only in one direction (communication is established but RTP > packages are only sent to the machine with SIPp client). SIP messages > between the boxes seems to arrive fine, and the communication after a while > closes without problem, telling the SIPp client that it was a Succesful > call. First I thought it could be Asterisk, but when I call from another > source (one more VoIP phone I have here) the communication is perfect and > RTP packages are running in both directions. So I think it can be SIPp. I > have used my own xml scenario based on uac_pcap.xml (it has little changes), > but even if I run the original uac_pcap, it doesn't work still. Anyway I add > my uac_pcap.xml: > > <?xml version="1.0" encoding="ISO-8859-1" ?> > <!DOCTYPE scenario SYSTEM "sipp.dtd"> > > <!-- This program is free software; you can redistribute it and/or --> > <!-- modify it under the terms of the GNU General Public License as --> > <!-- published by the Free Software Foundation; either version 2 of the --> > <!-- License, or (at your option) any later version. --> > <!-- --> > <!-- This program is distributed in the hope that it will be useful, --> > <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of --> > <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the --> > <!-- GNU General Public License for more details. --> > <!-- --> > <!-- You should have received a copy of the GNU General Public License --> > <!-- along with this program; if not, write to the --> > <!-- Free Software Foundation, Inc., --> > <!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA --> > <!-- --> > <!-- Sipp 'uac' scenario with pcap (rtp) play --> > <!-- --> > > <scenario name="UAC with media"> > <!-- In client mode (sipp placing calls), the Call-ID MUST be --> > <!-- generated by sipp. To do so, use [call_id] keyword. > --> > <send retrans="500" start_rtd="1"> > <![CDATA[ > > INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] > From: sipp <sip:sipp@ > [local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] > To: sut <sip:[EMAIL PROTECTED]:[remote_port]> > Call-ID: [call_id] > CSeq: 1 INVITE > Contact: sip:[EMAIL PROTECTED]:[local_port] > Max-Forwards: 70 > Subject: Performance Test > Content-Type: application/sdp > Content-Length: [len] > > v=0 > o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] > s=- > c=IN IP[local_ip_type] [local_ip] > t=0 0 > m=audio [auto_media_port] RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > ]]> > </send> > > <recv response="100" optional="true" rtd="1" start_rtd="2"> > </recv> > > <recv response="180" optional="true" rtd="2"> > </recv> > > <!-- 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 response="200" crlf="true"> > </recv> > > <!-- Packet lost can be simulated in any send/recv message by --> > <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. --> > <send> > <![CDATA[ > > ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] > From: sipp <sip:sipp@ > [local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] > To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param] > Call-ID: [call_id] > CSeq: 1 ACK > Contact: sip:[EMAIL PROTECTED]:[local_port] > Max-Forwards: 70 > Subject: Performance Test > Content-Length: 0 > > ]]> > </send> > > <!-- Play a pre-recorded PCAP file (RTP stream) --> > <nop> > <action> > <exec play_pcap_audio="pcap/call_g711a.pcap"/> > </action> > </nop> > > <!-- Pause 3 minutes, which is approximately the duration of the --> > <!-- PCAP file --> > <pause milliseconds="180000"/> > > <!-- Play an out of band DTMF '1' --> > <nop> > <action> > <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/> > </action> > </nop> > > <pause milliseconds="1000"/> > > <!-- The 'crlf' option inserts a blank line in the statistics report. --> > <send retrans="500"> > <![CDATA[ > > BYE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] > From: sipp <sip:sipp@ > [local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number] > To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param] > Call-ID: [call_id] > CSeq: 2 BYE > Contact: sip:[EMAIL PROTECTED]:[local_port] > Max-Forwards: 70 > Subject: Performance Test > Content-Length: 0 > > ]]> > </send> > > <recv response="200" crlf="true"> > </recv> > > <!-- definition of the response time repartition table (unit is ms) --> > <ResponseTimeRepartition value="50, 100, 200, 500, 1100, 2100, 3100, 4100, > 5100, 6100, 10000"/> > > <!-- definition of the call length repartition table (unit is ms) --> > <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> > > </scenario> > > > I am quite crazy about the whole problem, it makes me mad that it works > using two phones and that it doesn't when I generate the call with SIPp. > Maybe I have to add something to the SIP messages in the script that I am > not realizing or I don't know... > > Thanks, greetings, > Jes?s > > PS. I am not using NAT, all the machines and phones are in the same private > network. > -------------- next part -------------- > An HTML attachment was scrubbed... > > ------------------------------ > > ------------------------------------------------------------------------- > 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=100&url=/ > > ------------------------------ > > _______________________________________________ > Sipp-users mailing list > [email protected]<mailto:[email protected]> > https://lists.sourceforge.net/lists/listinfo/sipp-users > > > End of Sipp-users Digest, Vol 27, Issue 3 > ***************************************** > > > > -- > Darshan B N > > Thanks & Regards > > > > -- > Darshan B N > > Thanks & Regards > -- cheers!!!! sarvpriya ------------------------------------------------------------------------- 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=100&url=/ _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
