Hi. We use Sipp as a SIP generator for testing. We compile Sipp followed instruction: http://sipp.sourceforge.net/wiki/index.php/Compilation#Windows_XP <http://sipp.sourceforge.net/wiki/index.php/Compilation#Windows_XP>on Windows XP using cygwin using the command: "make pcapplay_ossl_cygwin ". with "make clean" before it. Compilation ends successfully but…
1. When we run the executable with pcap play nothing happens, no errors or printouts, it simply exits immediatly. 2. The executable size is only 900K where as the one on sipp site (version 3.1) is ~1.3mb. 3. On sipp.exe -v also nothing. Command that we use for running scenario: sipp -sf scen1.xml -inf inf3.csv -r XX -rp YY -l ZZ -i 192.168.1.x -p 5060 192.168.1.y:5060 (see attached) Compilation log attached. Can anyone help? Regards Vladimir
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<label id="0" />
<send retrans="500">
<![CDATA[
INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field0]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[field1]@[local_ip]:[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="503" rtd="true" next="1" optional="true" />
<recv response="400" rtd="true" next="1" optional="true" />
<recv response="403" rtd="true" next="1" optional="true" />
<recv response="404" rtd="true" next="1" optional="true" />
<recv response="603" rtd="true" next="1" optional="true" />
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" 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" next="2" 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. -->
<label id="2" />
<send>
<![CDATA[
ACK sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field0]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[field1]@[local_ip]:[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="7000"/>
<!--<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="7000"/>
<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="7000"/>
<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="7000"/>
<!-- 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="7000"/>
<!-- 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:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field0]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:[field1]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="503" rtd="true" next="1" optional="true" />
<recv response="400" rtd="true" next="1" optional="true" />
<recv response="403" rtd="true" next="1" optional="true" />
<recv response="404" rtd="true" next="1" optional="true" />
<recv response="603" rtd="true" next="1" optional="true" />
<recv response="200" rtd="true" next="3" crlf="true" />
<label id="1" />
<nop>
<action>
<exec int_cmd="stop_call" />
</action>
</nop>
<label id="3" />
<!-- 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>
inf3.csv
Description: inf3.csv
make.log
Description: make.log
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