[svn-commits] mmichelson: testsuite/asterisk/trunk r6054 - in /asterisk/trunk/tests/channel...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Dec 8 10:46:08 CST 2014
Author: mmichelson
Date: Mon Dec 8 10:46:05 2014
New Revision: 6054
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=6054
Log:
Add test to ensure that receiving a 491 in response to a reinvite does not cause a crash.
Added:
asterisk/trunk/tests/channels/pjsip/pending_reinvite/
asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/
asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/
asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/pending_reinvite/sipp/
asterisk/trunk/tests/channels/pjsip/pending_reinvite/sipp/uac-491.xml (with props)
asterisk/trunk/tests/channels/pjsip/pending_reinvite/test-config.yaml (with props)
Added: asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/extensions.conf?view=auto&rev=6054
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/extensions.conf Mon Dec 8 10:46:05 2014
@@ -1,0 +1,5 @@
+[default]
+exten => test,1,Answer()
+same => n,Set(CONNECTEDLINE(all)=TEST <1234>)
+same => n,Echo()
+same => n,Hangup()
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svn:keywords = 'Author Date Id Revision'
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Added: asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/pjsip.conf?view=auto&rev=6054
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/pending_reinvite/configs/ast1/pjsip.conf Mon Dec 8 10:46:05 2014
@@ -1,0 +1,9 @@
+[main-transport]
+type=transport
+protocol=udp
+bind=127.0.0.1
+
+[sipp]
+type=endpoint
+send_pai=yes
+allow=ulaw
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svn:keywords = 'Author Date Id Revision'
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Added: asterisk/trunk/tests/channels/pjsip/pending_reinvite/sipp/uac-491.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/pending_reinvite/sipp/uac-491.xml?view=auto&rev=6054
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/pending_reinvite/sipp/uac-491.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/pending_reinvite/sipp/uac-491.xml Mon Dec 8 10:46:05 2014
@@ -1,0 +1,134 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="491 Test">
+ <!-- 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: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:sipp@[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:8 PCMA/8000
+ a=rtpmap:0 PCMU/8000
+ a=ptime:20
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="181"
+ 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:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="INVITE" crlf="true" />
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 491 Request Pending
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+
+ </send>
+
+ <recv request="ACK" crlf = "true" />
+
+ <recv request="INVITE" crlf="true" />
+
+ <send retrans="500">
+ <![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>
+
+ <recv request="ACK" crlf = "true" />
+
+ <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:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:[service]@[remote_ip]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[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/trunk/tests/channels/pjsip/pending_reinvite/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/pending_reinvite/test-config.yaml?view=auto&rev=6054
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/pending_reinvite/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/pending_reinvite/test-config.yaml Mon Dec 8 10:46:05 2014
@@ -1,0 +1,35 @@
+testinfo:
+ summary: 'Ensure graceful operation when receiving a 491 response on a reinvite.'
+ description: |
+ 'A SIPp scenario places a call into Asterisk. Once the call has been answered, we
+ trigger a reinvite by calling the CONNECTEDLINE() dialplan function. The SIPp
+ scenario responds to the reinvite with a 491 response. Asterisk then waits a small
+ amount of time (approximately 2 seconds) and then sends a new reinvite. The SIPp
+ scenario responds this time with a 200 OK, and then hangs up the call.
+
+ The reason this test exists is because there was an issue in Asterisk 12+ for a time
+ where reception of a 491 response in Asterisk would result in a crash due to improper
+ reference counting of a PJSIP structure. This test ensures that the code changes have
+ fixed the issue as expected.'
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpTestCase'
+
+sipp-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'uac-491.xml'}}
+
+properties:
+ minversion: '12.8.0'
+ dependencies:
+ - python: 'twisted'
+ - python: 'starpy'
+ - app: 'sipp'
+ - asterisk: 'res_pjsip'
+ - asterisk: 'res_pjsip_session'
+ tags:
+ - pjsip
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