asterisk/res/pjproject/tests/pjsua/scripts-sipp/uac-reinvite-bad-via-branch.xml

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

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<!-- 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.                    -->
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<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                                                                    -->

<!-- Re-INVITE with bad Via branch (it has the same branch as the
     previous INVITE (ticket #965) will cause assertion
-->
     

<scenario name="UAC re-INVITE with bad Via branch">
  <send retrans="500">
    <![CDATA[

      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bKPj-1
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=Tester 234 123 IN IP4 127.0.0.1
      s=Tester
      c=IN IP4 127.0.0.1
      t=0 0
      m=audio 17424 RTP/AVP 0 101
      a=rtpmap:101 telephone-event/8000
      a=sendrecv

    ]]>
  </send>

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

  <recv response="180" optional="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.                             -->
  <recv response="200" rtd="true">
  </recv>

  <!-- 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:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bKPj-2
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>


  <!-- Re-INVITE with Via branch value the same as previous INVITE -->
  <send retrans="500">
    <![CDATA[

      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bKPj-1
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 INVITE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=Tester 234 124 IN IP4 127.0.0.1
      s=Tester
      c=IN IP4 127.0.0.1
      t=0 0
      m=audio 17424 RTP/AVP 0 101
      a=rtpmap:101 telephone-event/8000


    ]]>
  </send>

  <!-- 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.                             -->
  <recv response="500" rtd="true">
  </recv>

  <!-- 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:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bKPj-1
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>


  <pause milliseconds="2000"/>


  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  <send retrans="500">
    <![CDATA[

      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 3 BYE
      Contact: sip:sipp@[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>