[svn-commits] kmoore: testsuite/asterisk/trunk r4787 - in /asterisk/trunk/tests/channels/SI...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Mar 5 07:52:44 CST 2014
Author: kmoore
Date: Wed Mar 5 07:52:38 2014
New Revision: 4787
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4787
Log:
Testsuite: Add test for direct RTP reinvite failure
This adds a test for the scenario where Asterisk attempts to initiate a
remote RTP native bridge, but one side declines and hangs up. This
could previously cause a crash in Asterisk 1.8 and 11.
(closes issue ASTERISK-23310)
Review: https://reviewboard.asterisk.org/r/3297/
Added:
asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/
asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/
asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/
asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/
asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml (with props)
asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-reinvite.xml (with props)
asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/tests.yaml
Added: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/extensions.conf?view=auto&rev=4787
==============================================================================
--- asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/extensions.conf Wed Mar 5 07:52:38 2014
@@ -1,0 +1,4 @@
+[default]
+exten => s,1,NoOp()
+same => n,Dial(SIP/no_reinvite,,g)
+same => n,Playback(demo-instruct)
Propchange: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/extensions.conf
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Added: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf?view=auto&rev=4787
==============================================================================
--- asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf Wed Mar 5 07:52:38 2014
@@ -1,0 +1,12 @@
+[general]
+[no_reinvite]
+host=127.0.0.1
+port=5065
+type=friend
+insecure=invite
+
+[reinvite]
+host=127.0.0.1
+port=5066
+type=friend
+insecure=invite
Propchange: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf
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Propchange: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/configs/ast1/sip.conf
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svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml?view=auto&rev=4787
==============================================================================
--- asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml (added)
+++ asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml Wed Mar 5 07:52:38 2014
@@ -1,0 +1,108 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp default 'uas' scenario. -->
+<!-- -->
+
+<scenario name="UAS for native bridging preventing reinvites">
+ <Global variables="remote_tag" />
+ <recv request="INVITE" crlf="true">
+ <action>
+ <!-- Save the from tag. We'll need it when we send our BYE -->
+ <ereg regexp=".*(;tag=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag" />
+ </action>
+ </recv>
+
+ <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:[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 request="INVITE" crlf="true" />
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 500 Direct Media Reinvite Denied
+ [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-Type: application/sdp
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="ACK" rtd="true" crlf="true" />
+
+ <pause />
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: ua1 <sip:ua1@[local_ip]:[local_port]>;tag=[pid]SIPpTag01[call_number]
+ To: ua1 <sip:[service]@[remote_ip]:[remote_port]>[$remote_tag]
+ [last_Call-ID:]
+ CSeq: [cseq] BYE
+ Contact: sip:ua1@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Path Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
Propchange: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-no-reinvite.xml
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svn:mime-type = text/xml
Added: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-reinvite.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-reinvite.xml?view=auto&rev=4787
==============================================================================
--- asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-reinvite.xml (added)
+++ asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/sipp/uas-reinvite.xml Wed Mar 5 07:52:38 2014
@@ -1,0 +1,129 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp default 'uas' scenario. -->
+<!-- -->
+
+<scenario name="UAS for native bridging preventing reinvites">
+ <recv request="INVITE" crlf="true" />
+
+ <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:[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 request="INVITE" crlf="true" />
+
+ <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:[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 request="INVITE" crlf="true" />
+
+ <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:[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 request="BYE" rtd="true" crlf="true" />
+
+ <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:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
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svn:mime-type = text/xml
Added: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml?view=auto&rev=4787
==============================================================================
--- asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/test-config.yaml Wed Mar 5 07:52:38 2014
@@ -1,0 +1,40 @@
+testinfo:
+ summary: 'Basic direct media reinvite fallback test'
+ description: |
+ "This test verifies that Asterisk will fall back to P2P bridging if a direct media reinvite fails and checks for conditions where such an event fails to clean up properly."
+
+properties:
+ minversion: '1.8.27.0'
+ dependencies:
+ - python : 'twisted'
+ - python : 'starpy'
+ - app : 'sipp'
+ - asterisk : 'chan_sip'
+ tags:
+ - SIP
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ config-section: originator-config
+ typename: 'pluggable_modules.Originator'
+
+sipp-config:
+ reactor-timeout: 90
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'uas-no-reinvite.xml', '-p': '5065'} }
+ - { 'key-args': {'scenario': 'uas-reinvite.xml', '-p': '5066'},
+ 'ordered-args': ['-rtp_echo'] }
+
+originator-config:
+ channel: 'SIP/reinvite'
+ context: 'default'
+ exten: 's'
+ priority: '1'
+ trigger: 'ami_connect'
Propchange: asterisk/trunk/tests/channels/SIP/direct_rtp_fallback/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=4787&r1=4786&r2=4787
==============================================================================
--- asterisk/trunk/tests/channels/SIP/tests.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/tests.yaml Wed Mar 5 07:52:38 2014
@@ -64,3 +64,4 @@
- dir: 'register_forbidden_retry'
- test: 'outbound_register_from'
- test: 'outbound_reregister_from'
+ - test: 'direct_rtp_fallback'
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