[asterisk-commits] file: testsuite/asterisk/trunk r3537 - in /asterisk/trunk/tests/channels/SIP:...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sun Nov 25 16:19:56 CST 2012
Author: file
Date: Sun Nov 25 16:19:50 2012
New Revision: 3537
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3537
Log:
Add a test which confirms that SIP direct RTP setup works.
(closes issue ASTERISK-20520)
Added:
asterisk/trunk/tests/channels/SIP/directrtpsetup/
asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/
asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/
asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/
asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml (with props)
asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml (with props)
asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/tests.yaml
Added: asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf?view=auto&rev=3537
==============================================================================
--- asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf Sun Nov 25 16:19:50 2012
@@ -1,0 +1,2 @@
+[default]
+exten => test,1,Dial(SIP/phoneB)
Propchange: asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf
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Propchange: asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/extensions.conf
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svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf?view=auto&rev=3537
==============================================================================
--- asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf Sun Nov 25 16:19:50 2012
@@ -1,0 +1,20 @@
+[general]
+udpbindaddr=127.0.0.1
+allowguest=no
+directmedia=yes
+directrtpsetup=yes
+
+[phoneA]
+type=user
+insecure=invite,port
+disallow=all
+allow=ulaw
+host=127.0.0.1
+port=5061
+
+[phoneB]
+type=peer
+host=127.0.0.1
+port=5063
+disallow=all
+allow=ulaw
Propchange: asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf
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Propchange: asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/directrtpsetup/configs/ast1/sip.conf
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svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml?view=auto&rev=3537
==============================================================================
--- asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml (added)
+++ asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml Sun Nov 25 16:19:50 2012
@@ -1,0 +1,94 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test at 127.0.0.1:5060 SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test at 127.0.0.1:5060
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ User-Agent: Channel Param Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] 42.42.42.42
+ s=-
+ c=IN IP[media_ip_type] 42.42.42.42
+ t=0 0
+ m=audio 6000 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>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="c=IN IP4 42.42.42.42" search_in="body" check_it="true" assign_to="1"/>
+ </action>
+ </recv>
+
+ <Reference variables="1"/>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test at 127.0.0.1:5060 SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test at 127.0.0.1:5060>[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>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:test at 127.0.0.1:5060 SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test at 127.0.0.1:5060>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:kartoffelsalat@[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/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_A.xml
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svn:keywords = Author Date Id Revision
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Added: asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml?view=auto&rev=3537
==============================================================================
--- asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml (added)
+++ asterisk/trunk/tests/channels/SIP/directrtpsetup/sipp/phone_B.xml Sun Nov 25 16:19:50 2012
@@ -1,0 +1,82 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE with direct IP address of other sipp instance">
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp="c=IN IP4 42.42.42.42" search_in="body" check_it="true" assign_to="1"/>
+ </action>
+ </recv>
+
+ <Reference variables="1"/>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <pause milliseconds="1000"/>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=guest3 53655765 2353687637 IN IP[local_ip_type] 42.42.42.42
+ s=-
+ c=IN IP[media_ip_type] 42.42.42.42
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <!-- RECV ACK -->
+ <recv request="ACK"/>
+
+ <recv request="INVITE" optional="true"/>
+
+ <recv request="BYE"/>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+</scenario>
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Added: asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml?view=auto&rev=3537
==============================================================================
--- asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml Sun Nov 25 16:19:50 2012
@@ -1,0 +1,26 @@
+testinfo:
+ summary: 'Test the directrtpsetup option'
+ description: |
+ This tests whether the directrtpsetup option works or not.
+
+test-modules:
+ add-test-to-search-path: 'True'
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ fail-on-any: False
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'phone_B.xml', '-i': '127.0.0.1', '-p': '5063'} }
+ - { 'key-args': {'scenario': 'phone_A.xml', '-i': '127.0.0.1', '-p': '5061'} }
+
+properties:
+ minversion: '1.8.0.0'
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ tags:
+ - SIP
Propchange: asterisk/trunk/tests/channels/SIP/directrtpsetup/test-config.yaml
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svn:mime-type = text/plain
Modified: asterisk/trunk/tests/channels/SIP/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/tests.yaml?view=diff&rev=3537&r1=3536&r2=3537
==============================================================================
--- asterisk/trunk/tests/channels/SIP/tests.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/tests.yaml Sun Nov 25 16:19:50 2012
@@ -53,4 +53,6 @@
- test: 'subscribe'
- test: 'rfc2833_dtmf_detect'
- test: 'device_state_notification'
+ - test: 'directrtpsetup'
- test: 'session_timers_require'
+
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