Neil, I haven't ever used pcap, so I have no experience with your issue. The Aborted looks like SIPp might be getting a signal that is killing it. Try running it through gdb to see what is causing the failure.
Charles [EMAIL PROTECTED] wrote on 03/17/2008 04:18:29 AM: > Take 2 J > Hi I have been reading blogs and the website for a few days now and > I cant get an explanation of what I am doing wrong here : > All I do is take the uac_pcap xml file (uac with media) and try > extend it by adding a few extra dtmfs and pauses, but sipp keeps failing. > If I take out the pauses (except for the 8 second one for g711a) > then it works but too quickly, so I really need to use the pauses. > Please could you shed some light on the issue for me? My xml and > context explanation are below? Thanks! > Here is the command line, main screen and xml file. Sipp seems > to abort instantly, I am not sure what the problem is but when I > take the pauses out it works, but I obviously need the pauses > > > [COMMAND LINE] > sipp -sf /usr/local/bin/neil_uac_pcap.xml 10.42.101.253 -s 1300 -i > 10.42.101.253 -r 1 -rp 1000 -trace_err > > > [Main Screen] (please change your font to a fixed font like Courier > New if it is not lined up) > > > ------------------------------ Scenario Screen -------- [1-9]: > Change Screen -- > Call-rate(length) Port Total-time Total-calls Remote-host > 1.0(0 ms)/1.000s 5061 0.00 s 0 10.42.101.253:5060(UDP) > > 0 new calls during 0.003 s period 3 ms scheduler resolution > 0 calls (limit 33) Peak was 0 calls, after 0 s > 0 Running, 0 Paused, 0 Woken up > 0 out-of-call msg (discarded) > 1 open sockets > 0 Total RTP pckts sent 0.000 last period RTP rate (kB/s) > > Messages Retrans Timeout Unexpected-Msg > INVITE ----------> 0 0 0 > 100 <---------- 0 0 0 0 > 180 <---------- 0 0 0 0 > 200 <---------- E-RTD1 0 0 0 0 > > ACK ----------> 0 0 > [ NOP ] > Pause [ 8000ms] 0 0 > [ NOP ] > Pause [ 1000ms] 0 0 > [ NOP ] > Aborted > > > > [XML FILE] > > > <?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"> > <![CDATA[ > > INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0 > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] > From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[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 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11,16 > > ]]> > </send> > > <recv response="100" optional="true"> > </recv> > > <recv response="180" optional="true"> > </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" rtd="true" 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:[EMAIL PROTECTED]:[local_port]>;tag=[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/g711a.pcap"/> > </action> > </nop> > > <!-- Pause 8 seconds, which is approximately the duration of the --> > <!-- PCAP file --> > <pause milliseconds="8000"/> > > <!-- Play an out of band DTMF '1' --> > <nop> > <action> > <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/> > </action> > </nop> > > <pause milliseconds="1000"/> > > > > <!-- HERE IS WHERE MY OWN CODE [STARTS] --> > > <!-- Play an out of band DTMF '2' --> > <nop> > <action> > <exec play_pcap_audio="pcap/dtmf_2833_2.pcap"/> > </action> > </nop> > > <pause milliseconds="1000"/> > > > > <!-- Play an out of band DTMF '3' --> > <nop> > <action> > <exec play_pcap_audio="pcap/dtmf_2833_3.pcap"/> > </action> > </nop> > > <pause milliseconds="1000"/> > > > > <!-- HERE IS WHERE MY OWN CODE [ENDS] --> > > <!-- 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:[EMAIL PROTECTED]:[local_port]>;tag=[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="10, 20, 30, 40, 50, 100, 150, 200"/> > > <!-- definition of the call length repartition table (unit is ms) --> > <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> > > </scenario> > > > > > > > > From: Sejal Shah [mailto:[EMAIL PROTECTED] > Sent: Friday, March 14, 2008 11:18 PM > To: Steinbuch, Neil > Subject: Re: [Sipp-users] Sip Pause with Nop problem > > I can't see your XML file. > On Thu, Mar 13, 2008 at 3:31 AM, Steinbuch, Neil <[EMAIL PROTECTED] > > wrote: > Hi > > I have been reading blogs and the website for a few days now and I > cant get an explanation of what I am doing wrong here : > > All I do is take the uac_pcap xml file (uac with media) and try > extend it by adding a few extra dtmfs and pauses, but sipp keeps failing. > > If I take out the pauses (except for the 8 second one for g711a) > then it works but too quickly, so I really need to use the pauses. > > Please could you shed some light on the issue for me? My xml file isattached.. > > To read FirstRand Bank's Disclaimer for this email click on the > following address or copy into your Internet browser: > https://www.fnb.co.za/disclaimer.html > If you are unable to access the Disclaimer, send a blank e-mail to > [EMAIL PROTECTED] and we will send you a copy of the > Disclaimer. > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Sipp-users mailing list > [email protected] > https://lists.sourceforge.net/lists/listinfo/sipp-users > > To read FirstRand Bank's Disclaimer for this email click on the > following address or copy into your Internet browser: > https://www.fnb.co.za/disclaimer.html > If you are unable to access the Disclaimer, send a blank e-mail to > [EMAIL PROTECTED] and we will send you a copy of the > Disclaimer. > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Sipp-users mailing list > [email protected] > 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/vse0120000070mrt/direct/01/ _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
