[svn-commits] mmichelson: testsuite/asterisk/trunk r6364 - in /asterisk/trunk/tests/channel...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Jan 30 12:10:48 CST 2015
Author: mmichelson
Date: Fri Jan 30 12:10:39 2015
New Revision: 6364
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=6364
Log:
Add test to ensure no channel is leaked if we receive a REINVITE from a channel that performs a transfer.
This test has a SIPp scenario perform a blind transfer that fails. Upon receiving the failure notification,
the SIPp scenario sends a reinvite to Asterisk. The test ensures that this reinvite does not cause the
creation of a new channel by ensuring that the channel count after the test completes is zero.
Review: https://reviewboard.asterisk.org/r/4343
Added:
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/tests.yaml (with props)
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/sipp/
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/sipp/transferer.xml (with props)
asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/test-config.yaml (with props)
Added: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/tests.yaml?view=auto&rev=6364
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/tests.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/tests.yaml Fri Jan 30 12:10:39 2015
@@ -1,0 +1,3 @@
+# Enter tests here in the order they should be considered for execution:
+tests:
+ - test: 'transferer_reinvite'
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Added: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/extensions.conf?view=auto&rev=6364
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/extensions.conf Fri Jan 30 12:10:39 2015
@@ -1,0 +1,10 @@
+[default]
+
+exten => echo,1,Answer()
+same => n,Echo()
+same => n,Hangup()
+
+exten => call-sipp,1,Dial(PJSIP/sipp)
+same => n,Hangup()
+
+exten => bad_exten,1,NoOp()
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Added: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/pjsip.conf?view=auto&rev=6364
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/configs/ast1/pjsip.conf Fri Jan 30 12:10:39 2015
@@ -1,0 +1,14 @@
+[main-transport]
+type = transport
+protocol = udp
+bind = 127.0.0.1
+
+[sipp]
+type = endpoint
+context = default
+allow = ulaw
+aors = sipp
+
+[sipp]
+type = aor
+contact = sip:sipp at 127.0.0.1:5061
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Added: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/sipp/transferer.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/sipp/transferer.xml?view=auto&rev=6364
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/sipp/transferer.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/sipp/transferer.xml Fri Jan 30 12:10:39 2015
@@ -1,0 +1,185 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="transferer">
+ <Global variables="remote_tag" />
+ <recv request="INVITE" crlf="true">
+ <!-- Save the from tag. We'll need it when we send our BYE -->
+ <action>
+ <ereg regexp="(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag"/>
+ </action>
+ </recv>
+
+ <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]SIPpTag00[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 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <pause milliseconds="2000" />
+
+ <!-- Blind transfer this sucker to the transfer_target -->
+ <send retrans="500">
+ <![CDATA[
+
+ REFER sip:sipp@[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]:[remote_port]>[$remote_tag]
+ Call-ID: [call_id]
+ CSeq: [cseq] REFER
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Refer-To: sip:bad_exten@[remote_ip]:[remote_port];user=phone
+ Referred-By: sip:sipp@[local_ip]:[local_port]
+ Content-Length: 0
+
+ ]]>
+
+ </send>
+
+ <recv response="202" rtd="true">
+ </recv>
+
+ <!-- We should receive two NOTIFYs from Asterisk. One will be a 180 ringing sipfrag -->
+ <!-- and the other will be a 503 sipfrag -->
+ <recv request="NOTIFY" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ [last_Event:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="NOTIFY" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ [last_Event:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:sipp@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sip:sipp@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:sipp@[remote_ip]:[remote_port]>[$remote_tag]
+ Call-ID: [call_id]
+ CSeq: [cseq] 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:0 PCMU/8000
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:sipp@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port]
+ From: sip:sipp@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+ To: sut <sip:sipp@[remote_ip]:[remote_port]>[$remote_tag]
+ Contact: sip:[service]@[local_ip]:[local_port]
+ Call-ID: [call_id]
+ CSeq: [cseq] ACK
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE" timeout="5000"/>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ [last_Event:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+ ]]>
+ </send>
+</scenario>
+
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Added: asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/test-config.yaml?view=auto&rev=6364
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/transfers/blind_transfer/off_nominal/transferer_reinvite/test-config.yaml Fri Jan 30 12:10:39 2015
@@ -1,0 +1,51 @@
+testinfo:
+ summary: "Ensure that an ill-timed reinvite does not result in a stuck channel"
+ description: |
+ 'Asterisk originates a call to a Local channel that runs Echo. The other half of
+ the local channel is placed into the dialplan and calls a SIPp scenario. The SIPp
+ scenario answers the call and then performs a blind transfer to an extension that
+ immediately hangs up. The SIPp scenario, upon receiving a NOTIFY that the blind
+ transfer has failed, sends a reinvite to Asterisk. Asterisk should respond to this
+ reinvite with a 200 OK and then immediately send a BYE. The channel test condition
+ is used to ensure that there are no active channels once the test concludes.'
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: sipp.SIPpTestCase
+ modules:
+ -
+ config-section: originator-config
+ typename: pluggable_modules.Originator
+
+sipp-config:
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'transferer.xml', '-i': '127.0.0.1', '-p': '5061' }}
+
+originator-config:
+ channel: 'Local/echo at default'
+ context: 'default'
+ exten: 'call-sipp'
+ priority: '1'
+ trigger: 'scenario_start'
+ scenario-name: 'transferer.xml'
+
+
+properties:
+ minversion: '13.2.0'
+ dependencies:
+ - sipp:
+ version: 'v3.0'
+ - asterisk: 'res_pjsip'
+ - asterisk: 'res_pjsip_session'
+ - asterisk: 'res_pjsip_refer'
+ - asterisk: 'chan_pjsip'
+ - asterisk: 'app_dial'
+ - asterisk: 'app_echo'
+ tags:
+ - pjsip
+ testconditions:
+ -
+ name: 'channels'
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