<?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 '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: 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 IP[media_ip_type] [media_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 response="100"
	    crlf="true">
  </recv>

  <recv response="180"
	    crlf="true">
  </recv>

  <recv request="REFER" crlf="true">
  </recv>
  
  <nop>
    <action>
      <assignstr assign_to="lvia" value="[last_Via:]" />
    </action>
  </nop>

  <send>
	<![CDATA[

      SIP/2.0 202 Accepted
      [$lvia]
      From: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      To: <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      Call-ID: [call_id]
      CSeq: 111 REFER
      User-Agent: Unknown
      Supported: timer, path, replaces
      Content-Length: 0

	]]>
  </send>
  
  <pause milliseconds="1000"/>

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

      NOTIFY sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=cfcfcfcfcf
      Max-Forwards: 70
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 NOTIFY
      Contact: sip:sipp@[local_ip]:[local_port]
      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
      Supported: timer, path, replaces
      Event: refer;id=111
      Content-Type: message/sipfrag;version=2.0
      Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
      Subscription-State: terminated;reason=noresource
      Content-Length: 16

      SIP/2.0 200 Ok

	]]>
  </send>
  
  <recv response="200"
		crlf="true">
  </recv>
  
  <recv response="603"
		crlf="true">
  </recv>
  
  <send>
	<![CDATA[

      SIP/2.0 200 Ok
      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: sut <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 113 BYE
      User-Agent: Unknown
      Supported: timer, path, replaces
      Content-Length: 0

	]]>
  </send>

  <!-- 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>


