[svn-commits] kmoore: testsuite/asterisk/trunk r4690 - in /asterisk/trunk/tests/channels/pj...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Feb 12 07:43:29 CST 2014
Author: kmoore
Date: Wed Feb 12 07:43:22 2014
New Revision: 4690
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4690
Log:
Testsuite: Add basic NAT test
This adds a basic NAT test to ensure that ACK responses are sent to the
appropriate address even when the incoming Contact header contains a
private address.
(issue ASTERISK-23106)
Review: https://reviewboard.asterisk.org/r/3169/
Reported by: Matt Jordan
Added:
asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/
asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/
asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/
asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/sipp/
asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/sipp/ua1_invite_recv.xml (with props)
asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/tests.yaml
Added: asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/extensions.conf?view=auto&rev=4690
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/extensions.conf Wed Feb 12 07:43:22 2014
@@ -1,0 +1,6 @@
+[default]
+
+exten => s,1,NoOp()
+same => n,Answer()
+same => n,Dial(PJSIP/ua1)
+same => n,Hangup()
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Added: asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/pjsip.conf?view=auto&rev=4690
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/configs/ast1/pjsip.conf Wed Feb 12 07:43:22 2014
@@ -1,0 +1,18 @@
+[local-transport-udp]
+type=transport
+bind=127.0.0.1
+protocol=udp
+
+[ua1]
+type=endpoint
+aors=ua1
+context=default
+allow=!all,ulaw
+rewrite_contact=yes
+
+[ua1]
+type=aor
+max_contacts=1
+minimum_expiration=5
+default_expiration=30
+contact=sip:ua1 at 127.0.0.1:5062
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Added: asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/sipp/ua1_invite_recv.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/sipp/ua1_invite_recv.xml?view=auto&rev=4690
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/sipp/ua1_invite_recv.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/sipp/ua1_invite_recv.xml Wed Feb 12 07:43:22 2014
@@ -1,0 +1,89 @@
+<?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 NAT/contact rewrite testing">
+ <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:10.4.0.148:[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>
+
+ <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: [$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>
+
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Added: asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/test-config.yaml?view=auto&rev=4690
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/nat/test-config.yaml Wed Feb 12 07:43:22 2014
@@ -1,0 +1,33 @@
+testinfo:
+ summary: 'Verify contact rewriting functions properly for chan_pjsip'
+ description: |
+ 'This test verifies contact rewriting for chan_pjsip on outbound calls.'
+properties:
+ minversion: '12.1.0'
+ dependencies:
+ - python : 'twisted'
+ - python : 'starpy'
+ - app : 'sipp'
+ - asterisk : 'res_pjsip'
+ - asterisk : 'res_pjsip_nat'
+ tags:
+ - pjsip
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ config-section: originator-ua1_invite_recv
+ typename: 'pluggable_modules.Originator'
+
+originator-ua1_invite_recv:
+ trigger: 'ami_connect'
+
+sipp-config:
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'ua1_invite_recv.xml', '-p': '5062', '-send_timeout': '20000', '-recv_timeout': '20000', '-timeout': '25000'},}
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Modified: asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/tests.yaml?view=diff&rev=4690&r1=4689&r2=4690
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/tests.yaml (original)
+++ asterisk/trunk/tests/channels/pjsip/basic_calls/outgoing/nominal/tests.yaml Wed Feb 12 07:43:22 2014
@@ -2,3 +2,4 @@
- test: 'playback'
- test: 'echo'
- test: 'auth'
+ - test: 'nat'
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