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Viewing changes to daemon/libs/pjproject-2.2.1/tests/pjsua/scripts-sipp/strict-route.xml

  • Committer: Package Import Robot
  • Author(s): Jonathan Riddell
  • Date: 2015-01-07 14:51:16 UTC
  • mfrom: (4.3.5 sid)
  • Revision ID: package-import@ubuntu.com-20150107145116-yxnafinf4lrdvrmx
Tags: 1.4.1-0.1ubuntu1
* Merge with Debian, remaining changes:
 - Drop soprano, nepomuk build-dep
* Drop ubuntu patches, now upstream

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daemon/libs/pjproject-2.2.1/tests/pjsua/scripts-sipp/strict-route.xml
 
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<?xml version="1.0" encoding="ISO-8859-1" ?>
 
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<!DOCTYPE scenario SYSTEM "sipp.dtd">
 
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<!-- This program is free software; you can redistribute it and/or      -->
 
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<!-- modify it under the terms of the GNU General Public License as     -->
 
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<!-- published by the Free Software Foundation; either version 2 of the -->
 
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<!-- License, or (at your option) any later version.                    -->
 
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<!--                                                                    -->
 
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<!-- This program is distributed in the hope that it will be useful,    -->
 
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<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
 
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<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
 
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<!-- GNU General Public License for more details.                       -->
 
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<!--                                                                    -->
 
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<!-- You should have received a copy of the GNU General Public License  -->
 
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<!-- along with this program; if not, write to the                      -->
 
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<!-- Free Software Foundation, Inc.,                                    -->
 
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<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
 
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<!--                                                                    -->
 
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<!--                 Sipp default 'uas' scenario.                       -->
 
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<!--                                                                    -->
 
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<scenario name="Strict route test">
 
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  <recv request="INVITE" crlf="true">
 
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  </recv>
 
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  <send>
 
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    <![CDATA[
 
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      SIP/2.0 100 Trying
 
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      [last_Via:]
 
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      [last_From:]
 
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      [last_To:];tag=[call_number]
 
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      [last_Call-ID:]
 
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      [last_CSeq:]
 
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    ]]>
 
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  </send>
 
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  <send retrans="500">
 
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    <![CDATA[
 
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      SIP/2.0 407 Proxy Authenticate
 
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      [last_Via:]
 
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      [last_From:]
 
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      [last_To:];tag=[call_number]
 
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      [last_Call-ID:]
 
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      [last_CSeq:]
 
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      Proxy-Authenticate: DIGEST realm="test", nonce="12345", algorithm=MD5
 
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    ]]>
 
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  </send>
 
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  <recv request="ACK"
 
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        optional="false"
 
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        rtd="true"
 
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        crlf="true">
 
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  </recv>
 
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  <recv request="INVITE" crlf="true">
 
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  </recv>
 
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  <send>
 
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    <![CDATA[
 
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      SIP/2.0 100 Trying
 
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      [last_Via:]
 
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      [last_From:]
 
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      [last_To:];tag=[call_number]
 
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      [last_Call-ID:]
 
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      [last_CSeq:]
 
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    ]]>
 
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  </send>
 
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  <send>
 
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    <![CDATA[
 
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      SIP/2.0 180 Ringing
 
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      [last_Via:]
 
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      [last_From:]
 
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      [last_To:];tag=[call_number]
 
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      [last_Call-ID:]
 
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      [last_CSeq:]
 
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    ]]>
 
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  </send>
 
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  <send>
 
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    <![CDATA[
 
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      SIP/2.0 183 progress
 
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      [last_Via:]
 
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      [last_From:]
 
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      [last_To:];tag=[call_number]
 
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      [last_Call-ID:]
 
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      [last_CSeq:]
 
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      Contact: <sip:target@[local_ip]>
 
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      Record-route: <sip:proxy@[local_ip]:[local_port]>
 
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      Content-Type: application/sdp
 
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      v=0
 
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      o=- 3442013205 3442013205 IN IP4 [local_ip]
 
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      s=pjsip
 
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      c=IN IP4 [local_ip]
 
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      t=0 0
 
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      m=audio 4002 RTP/AVP 0
 
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      a=rtpmap:0 PCMU/8000
 
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    ]]>
 
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  </send>
 
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  <send retrans="500">
 
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    <![CDATA[
 
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      SIP/2.0 200 OK
 
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      [last_Via:]
 
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      [last_From:]
 
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      [last_To:];tag=[call_number]
 
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      [last_Call-ID:]
 
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      [last_CSeq:]
 
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      Contact: <sip:target@[local_ip]>
 
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      Record-route: <sip:proxy@[local_ip]:[local_port];maddr=[local_ip]>
 
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      Content-Type: application/sdp
 
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      v=0
 
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      o=- 3442013205 3442013205 IN IP4 [local_ip]
 
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      s=pjsip
 
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      c=IN IP4 [local_ip]
 
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      t=0 0
 
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      m=audio 4002 RTP/AVP 0
 
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      a=rtpmap:0 PCMU/8000
 
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    ]]>
 
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  </send>
 
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  <recv request="ACK"
 
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        optional="false"
 
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        rtd="true"
 
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        crlf="true">
 
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  </recv>
 
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  <recv request="BYE" crlf="true">
 
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  </recv>
 
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  <send>
 
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    <![CDATA[
 
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      SIP/2.0 200 OK
 
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      [last_Via:]
 
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      [last_From:]
 
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      [last_To:];tag=[call_number]
 
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      [last_Call-ID:]
 
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      [last_CSeq:]
 
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    ]]>
 
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  </send>
 
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  <!-- Keep the call open for a while in case the 200 is lost to be     -->
 
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  <!-- able to retransmit it if we receive the BYE again.               -->
 
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  <pause milliseconds="1000"/>
 
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  <!-- definition of the response time repartition table (unit is ms)   -->
 
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  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
 
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  <!-- definition of the call length repartition table (unit is ms)     -->
 
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  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
 
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</scenario>
 
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