[svn-commits] mmichelson: branch mmichelson/srv_sip_outbound r202 - in /asterisk/team/mmich...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Apr 6 10:19:15 CDT 2010
Author: mmichelson
Date: Tue Apr 6 10:19:13 2010
New Revision: 202
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=202
Log:
Initial add of second set of tests.
This has some sipp scenarios I'll be using plus the start of
an extensions.conf file to use. Next step will be adding
the test-config.yaml and test.lua files.
Added:
asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/
asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/configs/
asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/configs/extensions.conf (with props)
asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/
asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/dtmf_2833_1.pcap (with props)
asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/dtmf_2833_2.pcap (with props)
asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac.xml (with props)
asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas1.xml (with props)
asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas2.xml (with props)
Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/configs/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/configs/extensions.conf?view=auto&rev=202
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/configs/extensions.conf (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/configs/extensions.conf Tue Apr 6 10:19:13 2010
@@ -1,0 +1,9 @@
+[test_context]
+
+exten => test,1,Dial(SIP/peer//127.0.0.1:5061,,M(readdtmf),S(3))
+exten => test,n,Hangup
+
+[macro-readdtmf]
+
+exten => s,1,Read(READRESULT,,1,,1,5)
+exten => s,n,MacroExit
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Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/dtmf_2833_1.pcap
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/dtmf_2833_1.pcap?view=auto&rev=202
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Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/dtmf_2833_2.pcap
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/dtmf_2833_2.pcap?view=auto&rev=202
==============================================================================
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Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac.xml?view=auto&rev=202
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac.xml (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uac.xml Tue Apr 6 10:19:13 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: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: sut <sip:test@[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: sut <sip:[service]@[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: sipp <sip:caller@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: test <sip:test@[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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Added: asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas1.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas1.xml?view=auto&rev=202
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas1.xml (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas1.xml Tue Apr 6 10:19:13 2010
@@ -1,0 +1,125 @@
+<?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">
+ <!-- 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 request="INVITE" crlf="true">
+ </recv>
+
+ <!-- The '[last_*]' keyword is replaced automatically by the -->
+ <!-- specified header if it was present in the last message received -->
+ <!-- (except if it was a retransmission). If the header was not -->
+ <!-- present or if no message has been received, the '[last_*]' -->
+ <!-- keyword is discarded, and all bytes until the end of the line -->
+ <!-- are also discarded. -->
+ <!-- -->
+ <!-- If the specified header was present several times in the -->
+ <!-- message, all occurences are concatenated (CRLF seperated) -->
+ <!-- to be used in place of the '[last_*]' keyword. -->
+
+ <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 18
+ a=rtpmap:18 G729/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=sendrecv
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <nop>
+ <action>
+ <exec play_pcap_audio="sipp/dtmf_2833_1.pcap" />
+ </action>
+ </nop>
+
+ <pause milliseconds="200" />
+
+ <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>
+
+ <!-- Keep the call open for a while in case the 200 is lost to be -->
+ <!-- able to retransmit it if we receive the BYE again. -->
+ <pause 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/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas2.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas2.xml?view=auto&rev=202
==============================================================================
--- asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas2.xml (added)
+++ asterisk/team/mmichelson/srv_sip_outbound/tests/sip_outbound_address/sipp/uas2.xml Tue Apr 6 10:19:13 2010
@@ -1,0 +1,125 @@
+<?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">
+ <!-- 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 request="INVITE" crlf="true">
+ </recv>
+
+ <!-- The '[last_*]' keyword is replaced automatically by the -->
+ <!-- specified header if it was present in the last message received -->
+ <!-- (except if it was a retransmission). If the header was not -->
+ <!-- present or if no message has been received, the '[last_*]' -->
+ <!-- keyword is discarded, and all bytes until the end of the line -->
+ <!-- are also discarded. -->
+ <!-- -->
+ <!-- If the specified header was present several times in the -->
+ <!-- message, all occurences are concatenated (CRLF seperated) -->
+ <!-- to be used in place of the '[last_*]' keyword. -->
+
+ <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 18
+ a=rtpmap:18 G729/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-16
+ a=sendrecv
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <nop>
+ <action>
+ <exec play_pcap_audio="sipp/dtmf_2833_2.pcap" />
+ </action>
+ </nop>
+
+ <pause milliseconds="200" />
+
+ <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>
+
+ <!-- Keep the call open for a while in case the 200 is lost to be -->
+ <!-- able to retransmit it if we receive the BYE again. -->
+ <pause 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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