Hi, all
This below is the xml scenario file I'm using .
=================================================================
<?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: [field0] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1] <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="407" auth="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. -->
<!-- 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: [field0] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1] <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>
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: [field0] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1] <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:[EMAIL PROTECTED]:[local_port]
[field2]
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=-
t=0 0
c=IN IP[media_ip_type] [media_ip]
m=audio [auto_media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" rtd="true" crlf="true">
</recv>
<!-- 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="5000"/>
<!-- Play an out of band DTMF '1' '2' '3' '4' '5' '6' '7' '8' '9'
'*' -->
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
</action>
</nop>
<pause milliseconds="250"/>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_2.pcap"/>
</action>
</nop>
<pause milliseconds="250"/>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_3.pcap"/>
</action>
</nop>
<pause milliseconds="250"/>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_4.pcap"/>
</action>
</nop>
<pause milliseconds="250"/>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_5.pcap"/>
</action>
</nop>
<pause milliseconds="250"/>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_6.pcap"/>
</action>
</nop>
<pause milliseconds="250"/>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_7.pcap"/>
</action>
</nop>
<pause milliseconds="250"/>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_8.pcap"/>
</action>
</nop>
<pause milliseconds="250"/>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_9.pcap"/>
</action>
</nop>
<pause milliseconds="3000"/>
<!-- 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: [field0] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: [field1] <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>
==================================================================
I've installed sipp in windows, but when I use sipp with this file, it
told me that "The auto_media_port keyword re
quires PCAPPLAY."
What is PCAPPLAY ? I have installed winpcap already. And I alse give
the pcap file in the command option.
What else should I do to use sipp with this scenario file ?
Thanks so much.
Best Regards.
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