[asterisk-commits] mmichelson: branch mmichelson/srv_sip_outbound r207 - /asterisk/team/mmichels...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Apr 6 12:21:43 CDT 2010


Author: mmichelson
Date: Tue Apr  6 12:21:41 2010
New Revision: 207

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=207
Log:
Add more UAC scenarios.


Added:
    asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac1.xml   (with props)
    asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac2.xml   (with props)
    asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac3.xml   (with props)
    asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac4.xml   (with props)
    asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac5.xml   (with props)
    asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac6.xml   (with props)

Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac1.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac1.xml?view=auto&rev=207
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac1.xml (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac1.xml Tue Apr  6 12:21:41 2010
@@ -1,0 +1,118 @@
+<?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             -->
+<!--                                                                    -->
+<!--                 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:test1@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test1 <sip:test1@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:caller@[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[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <!-- 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 response="200" rtd="true">
+  </recv>
+
+  <!-- Packet lost can be simulated in any send/recv message by         -->
+  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
+  <send>
+    <![CDATA[
+
+      ACK sip:test1@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test1 <sip:test1@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- This delay can be customized by the -d command-line option       -->
+  <!-- or by adding a 'milliseconds = "value"' option here.             -->
+  <pause milliseconds="5000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:test1@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test1 <sip:test1@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <!-- 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>
+

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac1.xml
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac1.xml
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac1.xml
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac2.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac2.xml?view=auto&rev=207
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac2.xml (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac2.xml Tue Apr  6 12:21:41 2010
@@ -1,0 +1,118 @@
+<?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             -->
+<!--                                                                    -->
+<!--                 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:test2@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test2 <sip:test2@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:caller@[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[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <!-- 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 response="200" rtd="true">
+  </recv>
+
+  <!-- Packet lost can be simulated in any send/recv message by         -->
+  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
+  <send>
+    <![CDATA[
+
+      ACK sip:test2@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test2 <sip:test2@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- This delay can be customized by the -d command-line option       -->
+  <!-- or by adding a 'milliseconds = "value"' option here.             -->
+  <pause milliseconds="5000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:test2@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test2 <sip:test2@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <!-- 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>
+

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac2.xml
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac2.xml
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac2.xml
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac3.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac3.xml?view=auto&rev=207
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac3.xml (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac3.xml Tue Apr  6 12:21:41 2010
@@ -1,0 +1,118 @@
+<?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             -->
+<!--                                                                    -->
+<!--                 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:test3@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test3 <sip:test3@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:caller@[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[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <!-- 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 response="200" rtd="true">
+  </recv>
+
+  <!-- Packet lost can be simulated in any send/recv message by         -->
+  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
+  <send>
+    <![CDATA[
+
+      ACK sip:test3@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test3 <sip:test3@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- This delay can be customized by the -d command-line option       -->
+  <!-- or by adding a 'milliseconds = "value"' option here.             -->
+  <pause milliseconds="5000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:test@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test3 <sip:test3@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <!-- 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>
+

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac3.xml
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac3.xml
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac3.xml
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac4.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac4.xml?view=auto&rev=207
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac4.xml (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac4.xml Tue Apr  6 12:21:41 2010
@@ -1,0 +1,118 @@
+<?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             -->
+<!--                                                                    -->
+<!--                 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:test4@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test4 <sip:test4@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:caller@[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[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <!-- 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 response="200" rtd="true">
+  </recv>
+
+  <!-- Packet lost can be simulated in any send/recv message by         -->
+  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
+  <send>
+    <![CDATA[
+
+      ACK sip:test4@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test4 <sip:test4@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- This delay can be customized by the -d command-line option       -->
+  <!-- or by adding a 'milliseconds = "value"' option here.             -->
+  <pause milliseconds="5000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:test4@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test4 <sip:test4@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <!-- 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>
+

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac4.xml
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac4.xml
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac4.xml
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac5.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac5.xml?view=auto&rev=207
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac5.xml (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac5.xml Tue Apr  6 12:21:41 2010
@@ -1,0 +1,118 @@
+<?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             -->
+<!--                                                                    -->
+<!--                 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:test5@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test5 <sip:test5@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:caller@[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[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <!-- 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 response="200" rtd="true">
+  </recv>
+
+  <!-- Packet lost can be simulated in any send/recv message by         -->
+  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
+  <send>
+    <![CDATA[
+
+      ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test5 <sip:test5@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- This delay can be customized by the -d command-line option       -->
+  <!-- or by adding a 'milliseconds = "value"' option here.             -->
+  <pause milliseconds="5000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:test5@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test5 <sip:test5@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <!-- 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>
+

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac5.xml
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac5.xml
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac5.xml
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac6.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac6.xml?view=auto&rev=207
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac6.xml (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac6.xml Tue Apr  6 12:21:41 2010
@@ -1,0 +1,118 @@
+<?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             -->
+<!--                                                                    -->
+<!--                 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:test6@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test6 <sip:test6@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:caller@[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[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <!-- 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 response="200" rtd="true">
+  </recv>
+
+  <!-- Packet lost can be simulated in any send/recv message by         -->
+  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
+  <send>
+    <![CDATA[
+
+      ACK sip:test6@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test6 <sip:test6@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- This delay can be customized by the -d command-line option       -->
+  <!-- or by adding a 'milliseconds = "value"' option here.             -->
+  <pause milliseconds="5000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:test6@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: caller <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: test6 <sip:test6@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:caller@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <!-- 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>
+

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac6.xml
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac6.xml
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac6.xml
------------------------------------------------------------------------------
    svn:mime-type = text/plain




More information about the asterisk-commits mailing list