freeswitch/libs/unimrcp/tests/sipp/mrcp_uac_update

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


<scenario name="MRCP Synthesizer/Recognizer Resources UAC">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] token.                -->
  <send retrans="500">
    <![CDATA[

      INVITE 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]>
      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=user1 53655765 2353687637 IN IP4 [local_ip]
      s=-
      c=IN IP4 [media_ip]
      t=0 0
      m=application 9 TCP/MRCPv2 1
      a=setup:active
      a=connection:new
      a=resource:speechsynth
      a=cmid:1
      m=audio [media_port] RTP/AVP 0 8
      a=recvonly
      a=mid:1

	]]>
  </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=[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: 1 ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

  <!-- This delay can be customized by the -d command-line option       -->
  <!-- or by adding a 'milliseconds = "value"' option here.             -->
  <pause/>

  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] token.                -->
  <send retrans="500">
    <![CDATA[

      INVITE 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]>
      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=user1 53655765 2353687637 IN IP4 [local_ip]
      s=-
      c=IN IP4 [media_ip]
      t=0 0
      m=application 9 TCP/MRCPv2 1
      a=setup:active
      a=connection:new
      a=resource:speechsynth
      a=cmid:1
      m=audio [media_port] RTP/AVP 0 8
      a=sendrecv
      a=mid:1
      m=application 9 TCP/MRCPv2 1
      a=setup:active
      a=connection:existing
      a=resource:speechrecog
      a=cmid:1

	]]>
  </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=[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: 1 ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

  <!-- This delay can be customized by the -d command-line option       -->
  <!-- or by adding a 'milliseconds = "value"' option here.             -->
  <pause/>


  <!-- 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: 2 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>