If you need a thread per call, it is likely that there is an OS limit to the number of threads that you can use. I suggest trying to run multiple SIPp instances, and if that doesn't work you may possibly need to run SIPp on multiple machines. (Unless of course anyone else has a better explanation). You may also be able to adjust things with ulimit or sysctls.
Charles [EMAIL PROTECTED] wrote on 11/20/2007 08:00:42 AM: > I'm using sipp 2.0.1, compiled by hand with pcapplay support, on a > Debian 4.0 machine. At around 384 calls (plus or minus 1 or 2), sipp > consistently crashes with: > > 2007-11-20 13:41:39: Can create thread to send RTP packets. > sipp: There were more errors, enable -trace_err to log them. > > This occurs with the libpcap0.7-dev and libpcap0.8-dev Debian packages, > and also with libpcap-0.9.8 compiled from source. The command I'm using is: > > sipp -sf client.xml -s 1234567890 1.2.3.4 -r 1 -l 500 -rp 10 > > (IP address modified), and client.xml is attached. > > Can anyone shed any light on this? > > -- > Alistair Cunningham > +1 888 468 3111 > +44 20 799 39 799 > sip:[EMAIL PROTECTED] > http://integrics.com/ > <?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 [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) using lo interface --> > <!-- and that was originally meant for destination port 2006 --> > <nop> > <action> > <exec play_pcap_audio="g711a.pcap"/> > </action> > </nop> > > <!-- Pause 90 seconds, which is approximately the duration of the --> > <!-- PCAP file --> > <pause milliseconds="90000"/> > > <!-- 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> > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2005. > 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 2005. http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
