[svn-commits] mmichelson: branch mmichelson/phone-testsuite r3200 - in /asterisk/team/mmich...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Apr 19 16:22:46 CDT 2012
Author: mmichelson
Date: Thu Apr 19 16:22:40 2012
New Revision: 3200
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3200
Log:
Add tests for SIP custom INFO transmission.
Review: https://reviewboard.asterisk.org/r/1867
Added:
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/extensions.conf (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/sip.conf (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/run-test (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/excluded_useragent.xml (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/selected_useragent.xml (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/send_all.xml (with props)
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/test-config.yaml (with props)
Modified:
asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/tests.yaml
Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/extensions.conf?view=auto&rev=3200
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/extensions.conf (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/extensions.conf Thu Apr 19 16:22:40 2012
@@ -1,0 +1,10 @@
+[default]
+exten => _testX,1,Answer()
+exten => test1,2,SIPSendCustomINFO(sent to everyone!)
+exten => test2,2,SIPSendCustomINFO(sent to certain UAs,specificUA)
+exten => _testX,3,Hangup()
+[default]
+exten => _testX,1,Answer()
+exten => test1,2,SIPSendCustomINFO(sent to everyone!)
+exten => test2,2,SIPSendCustomINFO(sent to certain UAs,specificUA)
+exten => _testX,3,Hangup()
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Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/sip.conf?view=auto&rev=3200
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/sip.conf (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/configs/ast1/sip.conf Thu Apr 19 16:22:40 2012
@@ -1,0 +1,2 @@
+[general]
+[general]
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Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/run-test?view=auto&rev=3200
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/run-test (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/run-test Thu Apr 19 16:22:40 2012
@@ -1,0 +1,96 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2011, Digium, Inc.
+Kinsey Moore <kmoore at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+
+sys.path.append("lib/python")
+
+from twisted.internet import reactor
+from asterisk.sipp import SIPpTest
+
+
+WORKING_DIR = "SIP/custom_info"
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+SIPP_SCENARIOS = [
+ {'scenario' : 'send_all.xml',},
+ {'scenario' : 'selected_useragent.xml',},
+ {'scenario' : 'excluded_useragent.xml',},
+]
+
+# set port numberings
+port = 5061
+def update_entry(entry):
+ global port
+ entry['-p'] = "%d" % port
+ port += 1
+
+# generate SIPP scenarios with appropriate port numbers and the config to go with it
+def main():
+ [update_entry(i) for i in SIPP_SCENARIOS]
+ test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+ reactor.run()
+ if not test.passed:
+ return 1
+
+ return 0
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+#!/usr/bin/env python
+'''
+Copyright (C) 2011, Digium, Inc.
+Kinsey Moore <kmoore at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+
+sys.path.append("lib/python")
+
+from twisted.internet import reactor
+from asterisk.sipp import SIPpTest
+
+
+WORKING_DIR = "SIP/custom_info"
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+SIPP_SCENARIOS = [
+ {'scenario' : 'send_all.xml',},
+ {'scenario' : 'selected_useragent.xml',},
+ {'scenario' : 'excluded_useragent.xml',},
+]
+
+# set port numberings
+port = 5061
+def update_entry(entry):
+ global port
+ entry['-p'] = "%d" % port
+ port += 1
+
+# generate SIPP scenarios with appropriate port numbers and the config to go with it
+def main():
+ [update_entry(i) for i in SIPP_SCENARIOS]
+ test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+ reactor.run()
+ if not test.passed:
+ return 1
+
+ return 0
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
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Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/excluded_useragent.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/excluded_useragent.xml?view=auto&rev=3200
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/excluded_useragent.xml (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/excluded_useragent.xml Thu Apr 19 16:22:40 2012
@@ -1,0 +1,176 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <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: guest <sip:guest@[local_ip]:[local_port]>;tag=[call_number]
+ To: kartoffelsalat <sip:kartoffelsalat@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: excludedUA
+ 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 6000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" 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:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:guest0@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE"/>
+
+ <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:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <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: guest <sip:guest@[local_ip]:[local_port]>;tag=[call_number]
+ To: kartoffelsalat <sip:kartoffelsalat@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: excludedUA
+ 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 6000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" 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:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:guest0@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE"/>
+
+ <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:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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/phone-testsuite/tests/channels/SIP/custom_info/sipp/selected_useragent.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/selected_useragent.xml?view=auto&rev=3200
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/selected_useragent.xml (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/selected_useragent.xml Thu Apr 19 16:22:40 2012
@@ -1,0 +1,228 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <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: guest <sip:guest@[local_ip]:[local_port]>;tag=[call_number]
+ To: kartoffelsalat <sip:kartoffelsalat@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: specificUA
+ 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 6000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" 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:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:guest0@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="INFO" rtd="true">
+ <action>
+ <!-- make sure the body has the text we expect -->
+ <ereg regexp="sent to certain UAs"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE"/>
+
+ <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:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <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: guest <sip:guest@[local_ip]:[local_port]>;tag=[call_number]
+ To: kartoffelsalat <sip:kartoffelsalat@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: specificUA
+ 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 6000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" 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:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:guest0@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="INFO" rtd="true">
+ <action>
+ <!-- make sure the body has the text we expect -->
+ <ereg regexp="sent to certain UAs"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE"/>
+
+ <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:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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/phone-testsuite/tests/channels/SIP/custom_info/sipp/send_all.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/send_all.xml?view=auto&rev=3200
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/send_all.xml (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/send_all.xml Thu Apr 19 16:22:40 2012
@@ -1,0 +1,228 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <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: guest <sip:guest@[local_ip]:[local_port]>;tag=[call_number]
+ To: kartoffelsalat <sip:kartoffelsalat@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: specificUA
+ 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 6000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" 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:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:guest0@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="INFO" rtd="true">
+ <action>
+ <!-- make sure the body has the text we expect -->
+ <ereg regexp="sent to everyone"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE"/>
+
+ <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:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <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: guest <sip:guest@[local_ip]:[local_port]>;tag=[call_number]
+ To: kartoffelsalat <sip:kartoffelsalat@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: specificUA
+ 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 6000 RTP/AVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" 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:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:guest0@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="INFO" rtd="true">
+ <action>
+ <!-- make sure the body has the text we expect -->
+ <ereg regexp="sent to everyone"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: sip:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE"/>
+
+ <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:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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/phone-testsuite/tests/channels/SIP/custom_info/sipp/send_all.xml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/sipp/send_all.xml
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svn:mime-type = text/plain
Added: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/test-config.yaml?view=auto&rev=3200
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/test-config.yaml (added)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/test-config.yaml Thu Apr 19 16:22:40 2012
@@ -1,0 +1,22 @@
+testinfo:
+ summary: 'Test custom SIP INFO message generation.'
+ skip: 'Currently disabled until versioning can deal with digiumphones branches'
+ description: |
+ 'Ensure that custom SIP INFO messages are generated properly via AST_CONTROL_CUSTOM frames with AST_CUSTOM_SIP_INFO payload type.'
+
+properties:
+ dependencies:
+ - app : 'sipp'
+ tags:
+ - sip
+testinfo:
+ summary: 'Test custom SIP INFO message generation.'
+ skip: 'Currently disabled until versioning can deal with digiumphones branches'
+ description: |
+ 'Ensure that custom SIP INFO messages are generated properly via AST_CONTROL_CUSTOM frames with AST_CUSTOM_SIP_INFO payload type.'
+
+properties:
+ dependencies:
+ - app : 'sipp'
+ tags:
+ - sip
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/test-config.yaml
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Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/test-config.yaml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/custom_info/test-config.yaml
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svn:mime-type = text/plain
Modified: asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/tests.yaml?view=diff&rev=3200&r1=3199&r2=3200
==============================================================================
--- asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/tests.yaml (original)
+++ asterisk/team/mmichelson/phone-testsuite/tests/channels/SIP/tests.yaml Thu Apr 19 16:22:40 2012
@@ -41,3 +41,4 @@
- test: 'use_contact_from_200'
- test: 'generic_ccss'
- dir: 'sip_custom_presence'
+ - test: 'custom_info'
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