ims_msgc.xml

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

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<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--         IMS Benchmark Messaging scenario - UAC side                -->
<!--                                                                    -->
<!-- Author : David Verbeiren from Intel Corporation - July 2007        -->
<!--          Xavier Simonart from Intel Corporation - July 2007        -->
<!--          Philippe Lecluse from Intel Corporation - July 2007       -->
<!--                                                                    -->

<scenario name="ims_msgc">
  <info>
    <!-- The RTD metrics listed here are checked at the end of the call  -->
    <!-- In case one or more is exceeded, the call is marked as failed.  -->
    <metric ref="PX_TRT-PMM1" rtd="1" max="2000"/>
  </info>

<!-- *** STEP 1 *** -->
 <!-- This is a scenario preparation step. It involves:                        -->
 <!-- * Selecting a local user from a suitable pool                            -->
 <!-- * Requesting a suitable user from a partner SIPp and instructing it to   -->
 <!--   run the appropriate UAS side scenario to run against this one.         -->

  <nop>
    <action>
      <assign_user pool="2" scheme="rand_uni"/>  <!-- Select user from 'Registered' pool -->
      <move_user pool="4"/>                      <!-- So we don't take it again          -->
    </action>
  </nop>

  <sendRmt type="req_user">
    <param name="scenario" value="ims_msgs"/>
      <!-- The scenario that the partner must run.                                            -->
    <param name="from_uri" value="[field0]@[field1]"/>
      <!-- We must give the From URI (local user we selected) so that the partner SIPp can    -->
      <!-- detect when it later gets the SIP call that is prepared in this step.              -->
      <!-- Note: We can't use the call_id for this because the SUT could potentially change   -->
      <!-- the call_id between both call legs (SIPp-1 -> SUT -> SIPp-2)                       -->
    <param name="call_id" value="[call_id]"/>
      <!-- We include our local call_id so the partner SIPp will include it in any message it -->
      <!-- later sends to us (e.g. 'res_user' response below) thereby allowing us to          -->
      <!-- efficiently find back the call to which the message relates.                       -->
  </sendRmt>

  <recvRmt type="res_user" timeout="8000">
   <!-- When we get the response from the partner SIPp, we store the remote user URI in call  -->
   <!-- variables for later usage in the scenario (as 'To' user).                             -->
    <action>
      <store_param param="user_name" assign_to="1" />
      <store_param param="user_domain" assign_to="2" />
    </action>
  </recvRmt>

<!-- *** STEP 2 *** -->
 <!-- Now that our preparation steps are done, we wait for the time when the call must  -->
 <!-- actually start, according to the scenario arrival distribution.                   -->
  <sync crlf="true">
    <action>
      <exec int_cmd="set_start_time"/>  <!-- We remember our actual start time -->
    </action>
  </sync>

<!-- *** STEP 3 *** -->
 <!-- Now the SIP scenario really starts -->

  <send retrans="500" start_rtd="1" clen_random="uniform:0,140">
    <![CDATA[

      MESSAGE sip:[$1]@[$2] SIP/2.0
      Via: SIP/2.0/[transport] [field5]:[local_port];branch=[branch]
      Max-Forwards: 70
      Route: [$u1]
      From: "[field0]" <sip:[field0]@[field1]>;tag=[pid]SIPpTag00[call_number]
      To: "[$1]" <sip:[$1]@[$2]>
      Call-ID: [call_id]
      CSeq: 1 MESSAGE
      Contact: <sip:[field0]@[field5]:[local_port];transport=[transport]>
      Content-Type: text/plain
      Content-Length: [len]

      This is some text to appear in the instant message.
      It can have multiple lines and SIPp automatically computes the length to
      be put in Content-Length header.

    ]]>
  </send>

  <recv response="202" rtd="1">
    <action>
      <move_user pool="2"/>  <!-- Back to Registered pool !-->
    </action>
  </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>