[asterisk-commits] file: testsuite/asterisk/trunk r5970 - in /asterisk/trunk/tests/channels/pjsi...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Nov 20 09:11:46 CST 2014
Author: file
Date: Thu Nov 20 09:11:42 2014
New Revision: 5970
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=5970
Log:
Add a test for handling of an in-dialog INVITE with Replaces.
Added:
asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/
asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/
asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/
asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/sipp/
asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/sipp/invite_with_replaces.xml (with props)
asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/pjsip/tests.yaml
Added: asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/extensions.conf?view=auto&rev=5970
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/extensions.conf Thu Nov 20 09:11:42 2014
@@ -1,0 +1,4 @@
+[default]
+exten => playback,1,Answer()
+same => n,Playback(demo-congrats)
+same => n,Hangup()
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Added: asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/pjsip.conf?view=auto&rev=5970
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/configs/ast1/pjsip.conf Thu Nov 20 09:11:42 2014
@@ -1,0 +1,15 @@
+[local-transport-template](!)
+type=transport
+bind=127.0.0.1
+
+[local-transport-udp](local-transport-template)
+protocol=udp
+
+[endpoint-template-ipv4](!)
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+media_address=127.0.0.1
+
+[call](endpoint-template-ipv4)
+transport=local-transport-udp
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Added: asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/sipp/invite_with_replaces.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/sipp/invite_with_replaces.xml?view=auto&rev=5970
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/sipp/invite_with_replaces.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/sipp/invite_with_replaces.xml Thu Nov 20 09:11:42 2014
@@ -1,0 +1,129 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to playback with SDP in initial INVITE and then later an INVITE with Replaces">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:playback@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 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
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:playback@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[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:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:playback@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+ Replaces: <sip:taco@[remote_ip]>
+
+ v=0
+ o=phoneA 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
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:playback@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 ACK
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: 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>
+
+ <!-- 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/trunk/tests/channels/pjsip/in_dialog_invite_replaces/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/test-config.yaml?view=auto&rev=5970
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/in_dialog_invite_replaces/test-config.yaml Thu Nov 20 09:11:42 2014
@@ -1,0 +1,25 @@
+testinfo:
+ summary: 'Tests handling of an in-dialog INVITE with Replaces'
+ description: |
+ 'Run a SIPp scenario that sends a call to Asterisk and once answered sends an INVITE with Replaces in-dialog'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ fail-on-any: False
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'invite_with_replaces.xml', '-i': '127.0.0.1', '-p': '5061', '-d': '5000', '-s': 'call'} }
+
+properties:
+ minversion: '12.0.0'
+ dependencies:
+ - sipp :
+ version : 'v3.3'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
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Modified: asterisk/trunk/tests/channels/pjsip/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/tests.yaml?view=diff&rev=5970&r1=5969&r2=5970
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/tests.yaml (original)
+++ asterisk/trunk/tests/channels/pjsip/tests.yaml Thu Nov 20 09:11:42 2014
@@ -26,3 +26,4 @@
- test: 'accountcode'
- dir: 'publish'
- dir: 'optimistic_srtp'
+ - test: 'in_dialog_invite_replaces'
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