ims_uas.xml

<?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             -->
<!--                                                                    -->
<!--          IMS Benchmark Calling scenario - UAS 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_uas">

<!-- *** STEP 1 *** -->
 <!-- Wait for scenario to be initiated by a REQ_USER remote request which will specify   -->
 <!-- the ScenId -> UAS can then create a call with corresp scen and already feed it with -->
 <!-- the received REQ_USER message.                                                      -->

 <!-- Most of this this could possibly be fully hidden in the internal handling of        -->
 <!-- REQ_USER message that triggers an instance of the scenario but since we have the    -->
 <!-- necessary actions ('assign_user', 'move_user'), why not use them? (at the expense   -->
 <!-- of marginally lower perf than hard-coded behavior...)                               -->

  <recvRmt type="req_user">
    <action>
      <assign_user pool="2" scheme="rand_uni"/>
      <move_user pool="3"/>     <!-- So we don't take it again -->
    </action>
  </recvRmt>

  <sendRmt type="res_user" crlf="true">
    <param name="user_name" value="[field0]"/>
    <param name="user_domain" value="[field1]"/>
    <param name="call_id" value="[call_id]"/>
      <!-- We give our local call_id so the partner SIPp will include -->
      <!-- it in subsequent messages to us, allowing us to then       -->
      <!-- find the call back efficiently                             -->
  </sendRmt>

<!-- *** STEP 2 *** -->
 <!-- This is now the actual scenario. Since we are the controlled partner in this -->
 <!-- scenario (we got the 'req_user' request), our SIP scenario should start by a -->
 <!-- expecting an incoming SIP message                                            -->

  <recv request="INVITE" start_rtd="5,2">
   <!-- The imestamp, stored in rtd[2], will be reported to UAC side at end of -->
   <!-- scenario so it can compute the time the INVITE took from UAC to UAS.   -->
   <!-- rtd[5] measures the ringing time (between receiving INVITE and sending -->
   <!-- 200 OK.                                                                -->
    <action>
      <ereg regexp="(Via.*)([[:cntrl:]][[:cntrl:]]Via:[^[:cntrl:]]*)"
            search_in="msg"
            check_it="true"
            assign_to="1" />
      <ereg regexp="(Record-Route.*)([[:cntrl:]][[:cntrl:]]Record-Route:[^[:cntrl:]]*)"
            search_in="msg"
            check_it="true"
            assign_to="2" />
      <exec int_cmd="set_start_time"/>
    </action>
  </recv>

  <send>
    <![CDATA[

      SIP/2.0 183 Session Progress
      [last_Via:]
      [last_Record-Route:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[field0]@[field5]:[local_port];transport=[transport]>
      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 request="PRACK">
  </recv>

  <send>
  <![CDATA[
      
      SIP/2.0 200 OK
      [last_Via:]
      [last_Record-Route:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[field0]@[field5]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>


  <recv request="UPDATE">
  </recv>

  <send>
  <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_Record-Route:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[field0]@[field5]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>
  
  <!-- Ringing Time -->
 <pause poisson="true" mean="%RingTime"/>
 
  <send>
    <![CDATA[

      SIP/2.0 180 Ringing
      [$1]
      [$2]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      CSeq: 10 INVITE
      Contact: <sip:[field0]@[field5]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>

  <recv request="PRACK">
  </recv>

  <send>
  <![CDATA[
      
      SIP/2.0 200 OK
      [last_Via:]
      [last_Record-Route:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[field0]@[field5]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>

  <!-- Ringing Time -->
  <pause poisson="true" mean="%RingTime"/>

  <send retrans="500" rtd="5">
    <![CDATA[

      SIP/2.0 200 OK
      [$1]
      [$2]
      [routes]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      CSeq: 10 INVITE
      Contact: <sip:[field0]@[field5]:[local_port];transport=[transport]>
      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 request="BYE" optional="true" next="5"/>  <!-- In case it went faster than ACK (UDP) -->

  <recv request="ACK" start_rtd="1">
   <!-- Timestamp of ACK received is stored in rtd[1] and we send it to UAC at the -->
   <!-- end of the scenario so UAC can compute the full call establishment time    -->
   <!-- (from UAC sending INVITE to UAS receiving ACK, minus the ringing time)     -->
  </recv>

  <!-- Hold Time (call connected) - Controlled by UAC side in this scenario -->

  <recv request="BYE">
  </recv>

  <send crlf="true" next="10">
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Record-Route:]
      [last_Call-ID:]
      [last_CSeq:]
      Content-Length: 0

    ]]>
  </send>

  <label id="5"/> <!-- BYE came before ACK -->

  <send crlf="true">
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_Record-Route:]
      [last_CSeq:]
      Content-Length: 0

    ]]>
  </send>

  <recv request="ACK" optional="true" start_rtd="1"/>

  <label id="10"/> <!-- END OF SCENARIO -->

  <pause milliseconds="4000"/>
   <!-- Keep the call open for a while in case the 200 is lost to be     -->
   <!-- able to retransmit it if we receive the BYE again.               -->

<!-- *** STEP 3 *** -->
 <!-- Move the user back to a pool from which it will be picked for new scenarios -->
 <!-- and send to partner controlling the scenario the timing measurement we took -->
 <!-- so it can compute all timings and check IHS criterions.                     -->

  <nop>
    <action>
      <move_user pool="2"/>     <!-- Done with scen -> back to 'Registered' -->
    </action>
  </nop>

  <sendRmt type="res_call_info">
    <param name="rtd_info" rtd="1" info="timestamp"/>
     <!-- This sends the timestamp stored in start_time of rtd[1] -->
    <param name="rtd_info" rtd="2" info="timestamp"/>
    <param name="rtd_info" rtd="5" info="value"/>
     <!-- This sends the value (elapse time) that was measured by rtd[5] -->
  </sendRmt>

  <!-- 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>