ims_uac_full_ims.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 Calling 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_uac">
  <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-SES1" rtd="1" max="8000"/>
    <metric ref="PX_TRT-SES2" rtd="2" max="2000"/>
    <metric ref="PX_TRT-REL1" rtd="3" max="2000"/>
    <metric ref="CALL_DURATION" rtd="4"/>
  </info>

<!-- *** STEP 1 *** -->
 <!-- This is a scenario preparation step and is executed before the time the  -->
 <!-- scenario is scheduled to really start, so everything is ready (including -->
 <!-- at the partner SIPp - UAS - side) by then.                               -->
 <!-- This prepation 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="3"/>                      <!-- So we don't take it again          -->
    </action>
  </nop>

  <sendRmt type="req_user">
   <!-- The first 'sendRmt' command of a scenario, unless preceded by a 'recvRmt',            -->
   <!-- automatically selects a partner SIPp at random and remembers it for the duration of   -->
   <!-- the scenario.                                                                         -->
   <!-- The below parameters are to be encoded as IE (the mapping of param names to IEs is    -->
   <!-- done at scenario parsing time)                                                        -->
    <param name="scenario" value="ims_uas"/>
      <!-- The scenario that the partner must run. The scenario name given in 'value' will be -->
      <!-- looked up in the locally loaded scenarios and converted to a scenario id (int)     -->
      <!-- before sending the message to the partner. Hence it is required that all partners  -->
      <!-- have the exact same set of scenarios in the same order.                            -->
    <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 or 'res_call_info' later)        -->
      <!-- 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 random scenario arrival distribution.            -->
  <sync crlf="true">
    <action>
      <exec int_cmd="set_start_time"/>
    </action>
  </sync>

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

  <send retrans="500" start_rtd="1,2,4">  <!-- We start timer 1 and timer 2 -->
    <![CDATA[

      INVITE 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: 10 INVITE
      Contact: sip:[field0]@[field5]:[local_port]
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [field5]
      s=-
      c=IN IP[media_ip_type] [field5]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>

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

  <recv response="183" rrs="true">
  </recv>

   <send>
    <![CDATA[

      PRACK [next_url] SIP/2.0
      [last_Via:]
      Max-Forwards: 70
      [routes:]
      [last_From:]
      [last_To:]
      Call-ID: [call_id]
      CSeq: 11 PRACK
      Contact: sip:[field0]@[field5]:[local_port]
      Content-Length: 0

    ]]>
  </send>
  
  <recv response="200">
  </recv>

  <send>
    <![CDATA[

      UPDATE [next_url] SIP/2.0
      [last_Via:]
      Max-Forwards: 70
      [routes:]
      [last_From:]
      [last_To:]
      Call-ID: [call_id]
      CSeq: 12 UPDATE
      Contact: sip:[field0]@[field5]:[local_port]
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [field5]
      s=-
      c=IN IP[media_ip_type] [field5]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>

  <recv response="200">
  </recv>

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

  <send>
    <![CDATA[

      PRACK [next_url] SIP/2.0
      [last_Via:]
      Max-Forwards: 70
      [routes:]
      [last_From:]
      [last_To:]
      Call-ID: [call_id]
      CSeq: 13 PRACK
      Contact: sip:[field0]@[field5]:[local_port]
      Content-Length: 0

    ]]>
  </send>
 
  <recv response="200">
  </recv>


  <!-- By adding rrs="true" (Record Route Sets), the route sets are -->
  <!-- saved and used for following messages sent.                  -->
  <recv response="200" rrs="true">
  </recv>

  <send>
    <![CDATA[

      ACK [next_url] SIP/2.0
      [last_Via:]
      Max-Forwards: 70
      [routes:]
      From: "[field0]" <sip:[field0]@[field1]>;tag=[pid]SIPpTag00[call_number]
      [last_To:]
      Call-ID: [call_id]
      CSeq: 10 ACK
      Content-Length: 0

    ]]>
  </send>

  <!-- Hold Time (call connected) -->
  <!-- pause poisson="true" mean="%HoldTime"/ -->
  <pause exponential="true" mean="%HoldTime"/>

  <send retrans="500" start_rtd="3">
    <![CDATA[

      BYE [next_url] SIP/2.0
      [last_Via:]
      Max-Forwards: 70
      [routes:]
      From: "[field0]" <sip:[field0]@[field1]>;tag=[pid]SIPpTag00[call_number]
      [last_To:]
      Call-ID: [call_id]
      CSeq: 14 BYE
      Content-Length: 0

    ]]>
  </send>

  <recv response="200" crlf="true" rtd="3,4">
    <action>
      <move_user pool="2"/>  <!-- Back to Registered pool !-->
    </action>
  </recv>

<!-- *** STEP 4 *** -->
 <!-- After the scenario completes, we wait for timing measurements from the partner SIPp -->
  <recvRmt type="res_call_info" timeout="8000">
    <action>
      <rtd_eval rtd="2" start="2" stop="r2"/>
       <!-- This will look in the message recieved from partner for an IE     -->
       <!-- [rtd_info:[rtd_id:2][rtd_start_time:xyz]] and use that to compute -->
       <!-- the rtd (difference with the timestamp_rtd="2", so this measures  -->
       <!-- the time the INVITE took to reach the remote side).       -->
      <rtd_eval rtd="1" start="1" stop="r1"/>
       <!-- rtd[1] is between sending INVITE at UAC and receiving ACK at UAS. -->
       <!-- But we must substract ringing time from that:                     -->
      <rtd_store rtd="5" rmt_rtd="5"/>
       <!-- Store remote rtd[5] into local rtd[5]                             -->
      <rtd_op op="sub" rtd="1" rtd1="1" rtd2="5"/>
       <!-- Substract rtd[5] from rtd[1] and store the result into rtd[1].    -->
    </action>
  </recvRmt>

  <!-- 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>