[svn-commits] mmichelson: testsuite/asterisk/trunk r3347 - in /asterisk/trunk/contrib/sipp:...
SVN commits to the Digium repositories
svn-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>
+
Propchange: asterisk/trunk/contrib/sipp/calls/uac-hangup.xml
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svn:mime-type = text/plain
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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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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==============================================================================
--- 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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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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