Re: [Sipp-users] How to use sipp in windows ?

[Srivastava, A. K. <anu...@hp...>]

Hi,

This exe which was available unfortunately wasn't supporting pcap play.
You can try the latest exe available (3.1.1) .

Regards
Anuj

-----Original Message-----
From: sip...@li... [mailto:sip...@li...] On Behalf Of Andy
Sent: Friday, May 30, 2008 7:45 AM
To: Sip...@li...
Subject: [Sipp-users] How to use sipp in windows ?

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:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:[field0]@[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="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:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

  <send retrans="500">
  <![CDATA[

      INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 2 INVITE
      Contact: sip:[field0]@[local_ip]:[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:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 BYE
      Contact: sip:[field0]@[local_ip]:[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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