<?xml version="1.0" encoding="ISO-8859-1" ?>
<!--
-- Copyright (c) 2010-2022 Belledonne Communications SARL.
--
-- This file is part of Liblinphone 
-- (see https://gitlab.linphone.org/BC/public/liblinphone).
--
-- This program is free software: you can redistribute it and/or modify
-- it under the terms of the GNU Affero General Public License as
-- published by the Free Software Foundation, either version 3 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 Affero General Public License for more details.
--
-- You should have received a copy of the GNU Affero General Public License
-- along with this program. If not, see <http://www.gnu.org/licenses/>.
  -->
<!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             -->
<!--                                                                    -->
<!--                 Sipp default 'uas' scenario.                       -->
<!--                                                                    -->

<scenario name="Basic UAS responder">
  <!-- 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 request="INVITE" crlf="true">
  </recv>

  <!-- The '[last_*]' keyword is replaced automatically by the          -->
  <!-- specified header if it was present in the last message received  -->
  <!-- (except if it was a retransmission). If the header was not       -->
  <!-- present or if no message has been received, the '[last_*]'       -->
  <!-- keyword is discarded, and all bytes until the end of the line    -->
  <!-- are also discarded.                                              -->
  <!--                                                                  -->
  <!-- If the specified header was present several times in the         -->
  <!-- message, all occurences are concatenated (CRLF seperated)        -->
  <!-- to be used in place of the '[last_*]' keyword.                   -->

  <send>
    <![CDATA[

      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Content-Length: 0

    ]]>
  </send>

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

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 96 97 0 8 101 98
      a=rtpmap:96 speex/16000
      a=fmtp:96 vbr=on
      a=rtpmap:97 speex/8000
      a=fmtp:97 vbr=on
      a=rtpmap:101 telephone-event/16000
      a=rtpmap:98 telephone-event/8000

    ]]>
  </send>

  <recv request="ACK"
        optional="true"
        rtd="true"
        crlf="true">
  </recv>

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

  <send>
    <![CDATA[

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

    ]]>
  </send>

  <!-- 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.               -->
  <pause milliseconds="1000"/>


  <!-- 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>