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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!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             -->
<!--                                                                    -->
<!--        Based on sipp default 'uac' scenario.                       -->
<!-- To make one call to test@10.0.2.15 from IP 10.0.2.20 using PCMU codec:
     ./sipp -sf uac_media.xml 10.0.2.15 -i 10.0.2.20 -bind_local -m 1 -s test -key codec PCMU/8000 -key pt 99
  -->
<!--                                                                    -->

<scenario name="Codec test">
  <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: "[codec]" <sip:sipp@[local_ip]:[local_port]>;tag=[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: 70
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=- 42 42 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 [pt]
      a=rtpmap:[pt] [codec]
      a=recvonly
    ]]>
  </send>

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

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

  <recv response="183" optional="true">
  </recv>

  <recv response="200" rtd="true">
  </recv>

  <send>
    <![CDATA[
      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: "[codec]" <sip:sipp@[local_ip]:[local_port]>;tag=[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
      Content-Length: 0
    ]]>
  </send>

  <!-- wait for callee to hang up -->
  <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>
</scenario>