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Viewing changes to daemon/libs/pjproject-2.1.0/tests/pjsua/scripts-sipp/uac-ticket-1148.xml

  • Committer: Package Import Robot
  • Author(s): Francois Marier, Francois Marier, Mark Purcell
  • Date: 2014-10-18 15:08:50 UTC
  • mfrom: (1.1.12)
  • mto: This revision was merged to the branch mainline in revision 29.
  • Revision ID: package-import@ubuntu.com-20141018150850-2exfk34ckb15pcwi
Tags: 1.4.1-0.1
http://bugs.debian.org/759576
http://bugs.debian.org/741130
[ Francois Marier ]
* Non-maintainer upload
* New upstream release (closes: #759576, #741130)
  - debian/rules +PJPROJECT_VERSION := 2.2.1
  - add upstream patch to fix broken TLS support
  - add patch to fix pjproject regression

[ Mark Purcell ]
* Build-Depends:
  - sflphone-daemon + libavformat-dev, libavcodec-dev, libswscale-dev,
  libavdevice-dev, libavutil-dev
  - sflphone-gnome + libclutter-gtk-1.0-dev

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daemon/libs/pjproject-2.1.0/tests/pjsua/scripts-sipp/uac-ticket-1148.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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<scenario name="Ticket #1148 (assertion when offering SDP media with port zero but answered with port non-zero)">
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  <send retrans="500">
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    <![CDATA[
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      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
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      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
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      Call-ID: [call_id]
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      CSeq: 3 INVITE
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      Contact: sip:sipp@[local_ip]:[local_port]
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      Max-Forwards: 70
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      Subject: Performance Test
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      Content-Type: application/sdp
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      Content-Length: [len]
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      v=0
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      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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      s=-
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      c=IN IP[media_ip_type] [media_ip]
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      t=0 0
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      m=audio 4000 RTP/AVP 0
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      m=video 5000 RTP/AVP 100
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      a=rtpmap:100 H261/90000
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    ]]>
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  </send>
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  <recv response="100" optional="true">
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  </recv>
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  <recv response="180" optional="true">
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  </recv>
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  <recv response="200" rtd="true">
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  </recv>
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  <send>
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    <![CDATA[
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      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
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      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
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      Call-ID: [call_id]
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      CSeq: 3 ACK
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      Contact: sip:sipp@[local_ip]:[local_port]
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      Max-Forwards: 70
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      Subject: Performance Test
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      Content-Length: 0
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    ]]>
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  </send>
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  <!-- Waiting re-INVITE from pjsua  -->
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  <recv request="INVITE" crlf="true">
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  </recv>
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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:sipp@[local_ip]:[local_port]
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      Content-Type: application/sdp
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      v=0
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      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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      s=-
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      c=IN IP[media_ip_type] [media_ip]
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      t=0 0
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      m=audio 4000 RTP/AVP 0
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      m=video 5000 RTP/AVP 100
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      a=rtpmap:100 H261/90000
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    ]]>
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  </send>
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  <!-- Expecting assertion here -->
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  <recv request="ACK" crlf="true">
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  </recv>
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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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