[svn-commits] mmichelson: branch mmichelson/srv_sip_outbound r207 - /asterisk/team/mmichels...
SVN commits to the Digium repositories
svn-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
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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
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svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac2.xml
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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
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svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac3.xml
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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
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svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac4.xml
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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>
+
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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>
+
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