<?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 default 'branchc' scenario.                   -->
<!--                                                                    -->

<scenario name="branch_client">


 <send retrans>
    <![CDATA[

 INVITE sip:[field4]@[field2];user=phone SIP/2.0
 Via: SIP/2.0/[transport]  [local_ip]:[local_port];branch=[branch]
 From: "[field1]" <sip:+[field1]@[field2]>;tag=[call_number]
 To: <sip:[field4]@[field2];user=phone>
 Call-ID: [call_id]
 CSeq: 1 invite
 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, REGISTER
 Max-forwards: 70
 Contact: <sip:+[field1]@[local_ip]:[local_port];user=phone>
 Content-Type: application/sdp
 Content-Length: [len]

 v=0
      o=+[field1] 947638875 947638876 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 8 18 101
 a=rtpmap:8 PCMA/8000
 a=rtpmap:18 G729/8000
 a=cdsc:1 image udptl t38
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15
 a=sendrecv
 a=ptime:20

    ]]>
  </send>

  <recv response="100"  optional="true">
  </recv>

  <recv response="403" next="3" optional="true">
  </recv>

  <recv response="500" next="3" optional="true">
  </recv>
  <recv response="408" next="3" optional="true">
  </recv>
  <recv response="487" next="3" optional="true">
  </recv>

  <recv response="486" next="3" optional="true">
  </recv>

   <recv response="180" timeout="2000" >
    </recv>

    <recv response="200" rrs="true" timeout="1000" ontimeout="2">
    </recv>



<send>
    <![CDATA[

      ACK [next_url] SIP/2.0
      Via: SIP/2.0/[transport]  [local_ip]:[local_port];branch=[branch]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      CSeq: 1 ACK
      Content-Length: 0
 Max-Forwards: 70

    ]]>
  </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="8000"/>

  <!-- Play an out of band DTMF '1'                                     -->
  <nop>
    <action>
      <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
    </action>
  </nop>

  <pause milliseconds="1000"/>

<send>
    <![CDATA[

      BYE [next_url] SIP/2.0
      Via: SIP/2.0/[transport]  [local_ip]:[local_port];branch=[branch]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      CSeq: 2 BYE
      Content-Length: 0
 Max-Forwards: 70

    ]]>
  </send>


   <recv response="200" rrs="true">
  <action>
  <exec int_cmd="stop_call" />
  </action>
  </recv>


<label id="2"/>

<send>
    <![CDATA[

        CANCEL sip:[field4]@[field2];user=phone SIP/2.0
        Via: SIP/2.0/[transport]  [local_ip]:[local_port];branch=[branch]
        From: "[field1]" <sip:+[field1]@[field2]>;tag=[call_number]
        To: <sip:[field4]@[field2];user=phone>
        Call-ID: [call_id]
        CSeq: 1 CANCEL
        Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, REGISTER
        Max-forwards: 70
        Contact: <sip:+[field1]@[local_ip]:[local_port];user=phone>
        Content-Length: [len]
]]>
  </send>
<recv response="200" rrs="true">
   </recv>
<recv response="487" rrs="true">
   </recv>
<label id="3"/>
<send>
    <![CDATA[

      ACK sip:[field4]@[field2];user=phone SIP/2.0
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      CSeq: 1 ACK
      Content-Length: 0
        Max-Forwards: 70

    ]]>

  <action>
  <exec int_cmd="stop_call" />
  </action>

<!-- 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>

thanks in advance
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