[asterisk-commits] mmichelson: testsuite/asterisk/trunk r3347 - in /asterisk/trunk/contrib/sipp:...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Jul 20 10:40:29 CDT 2012


Author: mmichelson
Date: Fri Jul 20 10:40:26 2012
New Revision: 3347

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3347
Log:
Add some sample SIPp scenarios for people to use when constructing their own scenarios.

Review: https://reviewboard.asterisk.org/r/1929


Added:
    asterisk/trunk/contrib/sipp/
    asterisk/trunk/contrib/sipp/calls/
    asterisk/trunk/contrib/sipp/calls/uac-blind-transfer.xml   (with props)
    asterisk/trunk/contrib/sipp/calls/uac-hangup.xml   (with props)
    asterisk/trunk/contrib/sipp/calls/uac-no-hangup.xml   (with props)
    asterisk/trunk/contrib/sipp/calls/uas-blind-transfer.xml   (with props)
    asterisk/trunk/contrib/sipp/calls/uas-hangup.xml   (with props)
    asterisk/trunk/contrib/sipp/calls/uas-no-hangup.xml   (with props)
    asterisk/trunk/contrib/sipp/calls/uas-redirect.xml   (with props)
    asterisk/trunk/contrib/sipp/registration/
    asterisk/trunk/contrib/sipp/registration/uac-register.xml   (with props)
    asterisk/trunk/contrib/sipp/registration/uac-unregister.xml   (with props)
    asterisk/trunk/contrib/sipp/subscription/
    asterisk/trunk/contrib/sipp/subscription/uac-subscribe-no-unsubscribe.xml   (with props)
    asterisk/trunk/contrib/sipp/subscription/uac-subscribe-unsubscribe.xml   (with props)
    asterisk/trunk/contrib/sipp/table_of_contents   (with props)

Added: asterisk/trunk/contrib/sipp/calls/uac-blind-transfer.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/calls/uac-blind-transfer.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/calls/uac-blind-transfer.xml (added)
+++ asterisk/trunk/contrib/sipp/calls/uac-blind-transfer.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,155 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+  <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[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="181"
+        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: 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: 1 ACK
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause/>
+
+  <!-- Blind transfer this sucker to the transfer_target -->
+  <send retrans="500">
+    <![CDATA[
+
+      REFER 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]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 REFER
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Refer-To: sip:[transfer_target]@[remote_ip]:[remote_port];user=phone
+      Referred-By: sip:sipp@[local_ip]:[local_port]
+      Diversion: <sip:sipp@[local_ip]:[local_port]>;reason="send_to_vm"
+      Content-Length: 0
+
+    ]]>
+
+  </send>
+
+  <recv response="202" rtd="true">
+  </recv>
+
+  <!-- We should receive two NOTIFYs from Asterisk. One will be a 180 ringing sipfrag -->
+  <!-- and the other will be a 200 OK sipfrag -->
+  <recv request="NOTIFY" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      [last_Event:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <recv request="NOTIFY" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      [last_Event:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE 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]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: [cseq] BYE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+

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Added: asterisk/trunk/contrib/sipp/calls/uac-hangup.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/calls/uac-hangup.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/calls/uac-hangup.xml (added)
+++ asterisk/trunk/contrib/sipp/calls/uac-hangup.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,92 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+  <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[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="181"
+        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: 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: 1 ACK
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <pause/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE 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]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+

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Added: asterisk/trunk/contrib/sipp/calls/uac-no-hangup.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/calls/uac-no-hangup.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/calls/uac-no-hangup.xml (added)
+++ asterisk/trunk/contrib/sipp/calls/uac-no-hangup.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,90 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+  <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[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="181"
+        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: 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: 1 ACK
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <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>
+
+  <timewait milliseconds="4000"/>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+

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Added: asterisk/trunk/contrib/sipp/calls/uas-blind-transfer.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/calls/uas-blind-transfer.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/calls/uas-blind-transfer.xml (added)
+++ asterisk/trunk/contrib/sipp/calls/uas-blind-transfer.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,152 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+  <Global variables="remote_tag" />
+  <recv request="INVITE" crlf="true">
+      <!-- Save the from tag. We'll need it when we send our BYE -->
+      <action>
+          <ereg regexp="(;tag=.*)"
+              header="From:"
+              search_in="hdr"
+              check_it="true"
+              assign_to="remote_tag"/>
+	  </action>
+
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag00[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      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 request="ACK"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <pause/>
+
+  <!-- Blind transfer this sucker to the transfer_target -->
+  <send retrans="500">
+    <![CDATA[
+
+      REFER 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]>[$remote_tag]
+      Call-ID: [call_id]
+      CSeq: 2 REFER
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Refer-To: sip:[transfer_target]@[remote_ip]:[remote_port];user=phone
+      Referred-By: sip:sipp@[local_ip]:[local_port]
+      Diversion: <sip:sipp@[local_ip]:[local_port]>;reason="send_to_vm"
+      Content-Length: 0
+
+    ]]>
+
+  </send>
+
+  <recv response="202" rtd="true">
+  </recv>
+
+  <!-- We should receive two NOTIFYs from Asterisk. One will be a 180 ringing sipfrag -->
+  <!-- and the other will be a 200 OK sipfrag -->
+  <recv request="NOTIFY" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      [last_Event:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <recv request="NOTIFY" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      [last_Event:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE 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]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: [cseq] BYE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+

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Added: asterisk/trunk/contrib/sipp/calls/uas-hangup.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/calls/uas-hangup.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/calls/uas-hangup.xml (added)
+++ asterisk/trunk/contrib/sipp/calls/uas-hangup.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,106 @@
+<?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 'uas' scenario.                       -->
+<!--                                                                    -->
+
+<scenario name="Basic UAS responder">
+  <Global variables="remote_tag" />
+  <recv request="INVITE" crlf="true">
+      <!-- Save the from tag. We'll need it when we send our BYE -->
+      <action>
+          <ereg regexp=".*(;tag=.*)"
+              header="From:"
+              search_in="hdr"
+              check_it="true"
+              assign_to="remote_tag"/>
+	  </action>
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      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 request="ACK"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <pause/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE 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]SIPpTag01[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[$remote_tag]
+      [last_Call-ID:]
+      CSeq: [cseq] BYE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+

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Added: asterisk/trunk/contrib/sipp/calls/uas-no-hangup.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/calls/uas-no-hangup.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/calls/uas-no-hangup.xml (added)
+++ asterisk/trunk/contrib/sipp/calls/uas-no-hangup.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,78 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+  <recv request="INVITE" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      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 request="ACK"
+        optional="true"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <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>
+
+  <timewait milliseconds="4000"/>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+

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Added: asterisk/trunk/contrib/sipp/calls/uas-redirect.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/calls/uas-redirect.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/calls/uas-redirect.xml (added)
+++ asterisk/trunk/contrib/sipp/calls/uas-redirect.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,34 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic UAS responder">
+  <recv request="INVITE" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 302 Temporarily Moved
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[redir_target]@[remote_ip]:[remote_port];transport=[transport]>
+      Diversion: <sip:[local_ip]:[local_port]>;reason="send_to_vm"
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv request="ACK"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+

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Added: asterisk/trunk/contrib/sipp/registration/uac-register.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/registration/uac-register.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/registration/uac-register.xml (added)
+++ asterisk/trunk/contrib/sipp/registration/uac-register.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,23 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Basic Sipstone UAC">
+    <send retrans="500">
+        <![CDATA[
+
+        REGISTER 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: [aor]
+        Call-ID: [call_id]
+        CSeq: 1 REGISTER
+        Contact: [contact]
+        Max-Forwards: 70
+        Subject: Performance Test
+        Expires: [expires]
+        Content-Length: 0
+
+        ]]>
+    </send>
+
+    <recv response="200" rtd="true" />
+</scenario>

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Added: asterisk/trunk/contrib/sipp/registration/uac-unregister.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/registration/uac-unregister.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/registration/uac-unregister.xml (added)
+++ asterisk/trunk/contrib/sipp/registration/uac-unregister.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,23 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+
+<scenario name="Basic Sipstone UAC">
+    <send retrans="500">
+        <![CDATA[
+
+        REGISTER 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: [aor]
+        Call-ID: [call_id]
+        CSeq: 1 REGISTER
+        Contact: [contact]
+        Max-Forwards: 70
+        Subject: Performance Test
+        Expires: 0
+        Content-Length: 0
+
+        ]]>
+    </send>
+
+    <recv response="200" rtd="true" />
+</scenario>

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Added: asterisk/trunk/contrib/sipp/subscription/uac-subscribe-no-unsubscribe.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/subscription/uac-subscribe-no-unsubscribe.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/subscription/uac-subscribe-no-unsubscribe.xml (added)
+++ asterisk/trunk/contrib/sipp/subscription/uac-subscribe-no-unsubscribe.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,99 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+  <send retrans="500">
+    <![CDATA[
+
+      SUBSCRIBE 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 SUBSCRIBE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Event: [event]
+      Accept: [accept]
+      Expires: [expires]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <!-- Initial NOTIFY upon subscribing -->
+  <recv request="NOTIFY" crlf="true" />
+
+  <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>
+
+  <!-- Insert NOTIFY/200 OK pairs as needed for your test. An example is included -->
+  <!-- below, commented out. The example does not verify the body of the NOTIFY request -->
+
+  <!--
+  <recv request="NOTIFY" crlf="true" />
+
+  <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>
+  -->
+
+  <!-- NOTIFY terminating subscription -->
+  <recv request="NOTIFY" crlf="true" />
+
+  <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>
+
+  <timewait milliseconds="4000"/>
+
+  <!-- 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>
+

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Added: asterisk/trunk/contrib/sipp/subscription/uac-subscribe-unsubscribe.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/subscription/uac-subscribe-unsubscribe.xml?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/subscription/uac-subscribe-unsubscribe.xml (added)
+++ asterisk/trunk/contrib/sipp/subscription/uac-subscribe-unsubscribe.xml Fri Jul 20 10:40:26 2012
@@ -1,0 +1,121 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Basic Sipstone UAC">
+  <send retrans="500">
+    <![CDATA[
+
+      SUBSCRIBE 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 SUBSCRIBE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Event: [event]
+      Accept: [accept]
+      Expires: [expires]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <!-- Initial NOTIFY upon subscribing -->
+  <recv request="NOTIFY" crlf="true" />
+
+  <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>
+
+  <!-- Insert NOTIFY/200 OK pairs as needed for your test. An example is included -->
+  <!-- below, commented out. The example does not verify the body of the NOTIFY request -->
+
+  <!--
+  <recv request="NOTIFY" crlf="true" />
+
+  <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>
+  -->
+
+  <send retrans="500">
+    <![CDATA[
+
+      SUBSCRIBE 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]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: [cseq] SUBSCRIBE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Event: [event]
+      Accept: [accept]
+      Expires: 0
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true" />
+
+  <!-- NOTIFY terminating subscription -->
+  <recv request="NOTIFY" crlf="true" />
+
+  <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>
+
+  <timewait milliseconds="4000"/>
+
+  <!-- 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>
+

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Added: asterisk/trunk/contrib/sipp/table_of_contents
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/contrib/sipp/table_of_contents?view=auto&rev=3347
==============================================================================
--- asterisk/trunk/contrib/sipp/table_of_contents (added)
+++ asterisk/trunk/contrib/sipp/table_of_contents Fri Jul 20 10:40:26 2012
@@ -1,0 +1,102 @@
+Included here are common SIPp scenarios used for testing
+Asterisk. The scenarios are grouped based on their nature.
+
+Each scenario's name starts with either "uas" or "uac" to
+indicate the role the scenario plays.
+
+Below are descriptions of each of the scenarios.
+
+=========================================================
+|                         CALLS                         |
+=========================================================
+
+* uac-hangup.xml
+	This scenario closely mirrors the default UAC scenario
+	that SIPp uses. The only modification that has been
+	made is to optionally accept a 181 response during
+	call setup. The scenario calls a destination, waits
+	a set amount of time and then sends a BYE. The
+	destination can be specified with the -s option, and
+	the waiting period can be specified with the -d option.
+
+* uac-no-hangup.xml
+	This scenario is identical to uac-hangup.xml except
+	that it does not initiate the hangup. Instead, it waits
+	for a BYE to be sent to it, therefore the -d option
+	will have no effect when this scenario runs.
+
+* uac-blind-transfer.xml
+	This scenario establishes a dialog, waits a period of
+	time and then sends a REFER to blind transfer the call.
+	The scenario expects to receive a 202 response to the
+	REFER. After this, it expects two NOTIFY requests, to
+	which it responds with 200 OKs. After this, the
+	scenario sends a BYE. The -s option can be used to
+	control the destination of the original INVITE. The
+	-d option can be used to control the waiting period.
+	Use the -key option to set the "transfer_target"
+	to the destination of the transfer.
+
+* uas-no-hangup.xml
+	This scenario is the default SIPp UAS sceanrio. It
+	answers and incoming INVITE with a 200 OK, awaits an
+	ACK, and then awaits a BYE.
+
+* uas-hangup.xml
+	This scenario is identical to uas-no-hangup.xml except
+	that it sends a BYE instead of waiting for one. After
+	the initial INVITE transaction completes, the scenario

[... 67 lines stripped ...]



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