Charles,

"ulimit -s unlimited" cured the problem. I was thinking in terms of a 
thread limit rather than a stack limit.

Alistair Cunningham
+1 888 468 3111
+44 20 799 39 799
sip:[EMAIL PROTECTED]
http://integrics.com/


Charles P Wright wrote:
> 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>
>>
>>
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> 
> 

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