uac.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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<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
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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             -->
<!--                                                                    -->
<!--                 Sipp default 'uac' scenario.                       -->
<!--                                                                    -->

<scenario name="Basic Sipstone UAC">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
  <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=[pid]SIPpTag00[call_number]
      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 68
      Recv-Info: 
      Accept: application/media_control+xml, application/sdp
      MIME-Version: 1.0
      Content-Type: multipart/mixed;boundary=UniqueBroadWorksBoundary

      --UniqueBroadWorksBoundary
      Content-Type:application/rs-metadata+xml
      Content-Disposition:recording-session
      Content-Length:2493

      <?xml version="1.0" encoding="UTF-8"?>
      <recording_metadata xmlns="urn:ietf:params:xml:ns:siprec">
          <dataMode>complete</dataMode>
          <recording id="urn:uuid:e9755e57-aaf0-40fe-b3e3-9b54cfbfacc3">
              <requestor>SRC</requestor>
              <type>selective</type>
          </recording>
          <group id="urn:uuid:eb76d96b-3c33-48d2-b552-f2c2175e09e3" recording="urn:uuid:e9755e57-aaf0-40fe-b3e3-9b54cfbfacc3">
              <initiator>sip:+42191215@A_IP;user=phone</initiator>
          </group>
          <session id="urn:uuid:3e22dcc3-b01e-4c09-8fc8-8f49c0d43cac" group="urn:uuid:eb76d96b-3c33-48d2-b552-f2c2175e09e3">
              <start-time>2021-09-07T09:36:14+02:00</start-time>
          </session>
          <participant id="urn:uuid:3e954e94-e1e4-4699-b85b-cdad3f810ce6" session="urn:uuid:3e22dcc3-b01e-4c09-8fc8-8f49c0d43cac">
              <aor>sip:+42191215@A_IP;user=phone</aor>
              <send>
                  <id>urn:uuid:c48fc5cd-d40e-4bac-a76c-661d53a59250</id>
              </send>
          </participant>
          <participant id="urn:uuid:a13113cb-9eb3-48d8-bf74-1df3bf7837bc" session="urn:uuid:3e22dcc3-b01e-4c09-8fc8-8f49c0d43cac">
              <aor>sip:+48511325595@B_IP;user=phone</aor>
              <send>
                  <id>urn:uuid:f12c1e7c-cf4a-4ebb-8a96-caf32951c5ea</id>
              </send>
          </participant>
          <stream id="urn:uuid:c48fc5cd-d40e-4bac-a76c-661d53a59250" session="urn:uuid:3e22dcc3-b01e-4c09-8fc8-8f49c0d43cac">
              <label>1</label>
              <mode>separate</mode>
          </stream>
          <stream id="urn:uuid:f12c1e7c-cf4a-4ebb-8a96-caf32951c5ea" session="urn:uuid:3e22dcc3-b01e-4c09-8fc8-8f49c0d43cac">
              <label>2</label>
              <mode>separate</mode>
          </stream>
          <extensiondata id="urn:uuid:a469d484-56b8-43ec-b49e-d6339fc122a4" parent="urn:uuid:3e22dcc3-b01e-4c09-8fc8-8f49c0d43cac">
              <broadWorksRecordingMetadata version="1.0" xmlns="http://schema.broadsoft.com/broadworksCallRecording">
          <extTrackingID>8e71e709-9795-49cd-adf4-2e3f5696f485</extTrackingID>
          <serviceProviderID>JTENDOXXE1</serviceProviderID>
          <groupID>JTENDOE1G1_grp</groupID>
          <userID>42191215</userID>
          <callID>zaxLnjq6iRWVp3gWirrN_g</callID>
          <callType>
              <origCall>
                  <callingPartyNumber>sip:+42191215@A_IP;user=phone</callingPartyNumber>
                  <calledPartyNumber>sip:+48511325595@B_IP;user=phone</calledPartyNumber>
              </origCall>
          </callType>
          <recordingType>on</recordingType>
      </broadWorksRecordingMetadata>
          </extensiondata>
      </recording_metadata>

      --UniqueBroadWorksBoundary
      Content-Type: application/sdp
      Content-Length: 417

      v=0
      o=- 426088 426088 IN IP4 BW_SBC
      s=media server session
      b=AS:80
      t=0 0
      m=audio 23696 RTP/AVP 8
      c=IN IP4 BW_SBC
      b=AS:80
      b=RR:6000
      b=RS:2000
      a=rtpmap:8 PCMA/8000
      a=rtcp-xr
      a=ptime:20
      a=maxptime:80
      a=sendonly
      a=label:1
      m=audio 21080 RTP/AVP 8
      c=IN IP4 BW_SBC
      b=AS:80
      b=RR:6000
      b=RS:2000
      a=rtpmap:8 PCMA/8000
      a=rtcp-xr
      a=ptime:20
      a=maxptime:80
      a=sendonly
      a=label:2

      --UniqueBroadWorksBoundary--

    ]]>
  </send>

  <recv response="100"
        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=[pid]SIPpTag00[call_number]
      To: [service] <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=[pid]SIPpTag00[call_number]
      To: [service] <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>