[svn-commits] coreyfarrell: testsuite/asterisk/trunk r4681 - in /asterisk/trunk/tests/chann...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Feb 10 12:25:13 CST 2014
Author: coreyfarrell
Date: Mon Feb 10 12:25:08 2014
New Revision: 4681
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=4681
Log:
Create a basic test for chan_sip Record-Record handling.
Test strict and loose routes, for inbound and outbound calls. This
verifies expected method URL's, as well as Record-Route and Route headers.
(issue ASTERISK-22582)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3172/
Added:
asterisk/trunk/tests/channels/SIP/route/
asterisk/trunk/tests/channels/SIP/route/configs/
asterisk/trunk/tests/channels/SIP/route/configs/ast1/
asterisk/trunk/tests/channels/SIP/route/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/route/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/route/sipp/
asterisk/trunk/tests/channels/SIP/route/sipp/recv-invite.xml (with props)
asterisk/trunk/tests/channels/SIP/route/sipp/send-invite.xml (with props)
asterisk/trunk/tests/channels/SIP/route/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/tests.yaml
Added: asterisk/trunk/tests/channels/SIP/route/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/route/configs/ast1/extensions.conf?view=auto&rev=4681
==============================================================================
--- asterisk/trunk/tests/channels/SIP/route/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/route/configs/ast1/extensions.conf Mon Feb 10 12:25:08 2014
@@ -1,0 +1,8 @@
+[default]
+exten => s,1,Answer()
+same => n,Dial(SIP/ua1,1)
+same => n,Hangup()
+
+exten => _XX,1,Ringing()
+same => n,Answer()
+same => n,Hangup()
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Added: asterisk/trunk/tests/channels/SIP/route/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/route/configs/ast1/sip.conf?view=auto&rev=4681
==============================================================================
--- asterisk/trunk/tests/channels/SIP/route/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/route/configs/ast1/sip.conf Mon Feb 10 12:25:08 2014
@@ -1,0 +1,7 @@
+[general]
+udpbindaddr=127.0.0.1
+
+[ua1]
+type=peer
+host=127.0.0.1
+port=5062
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Added: asterisk/trunk/tests/channels/SIP/route/sipp/recv-invite.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/route/sipp/recv-invite.xml?view=auto&rev=4681
==============================================================================
--- asterisk/trunk/tests/channels/SIP/route/sipp/recv-invite.xml (added)
+++ asterisk/trunk/tests/channels/SIP/route/sipp/recv-invite.xml Mon Feb 10 12:25:08 2014
@@ -1,0 +1,146 @@
+<?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 -->
+
+<scenario name="recv-invite">
+ <nop hide="true">
+ <action>
+ <assignstr assign_to="1" value="[call_number]" />
+ <strcmp assign_to="rr_type" variable="1" value="1" />
+ </action>
+ </nop>
+
+ <nop hide="true" next="4" test="rr_type">
+ <action>
+ <assignstr assign_to="rr_out" value="<test1-subsystem1@[local_ip]:[local_port]>,<test1-subsystem2@[local_ip]:[local_port]>" />
+ <assignstr assign_to="rr_uri" value="test1-subsystem2@[local_ip]:[local_port]" />
+ <assignstr assign_to="rr_rcv" value="<test1-subsystem1@[local_ip]:[local_port]>,<sip:[local_ip]:[local_port]>" />
+ </action>
+ </nop>
+
+ <nop hide="true">
+ <action>
+ <assignstr assign_to="rr_out" value="<test1-subsystem1@[local_ip]:[local_port];lr>,<test1-subsystem2@[local_ip]:[local_port];lr>" />
+ <assignstr assign_to="rr_uri" value="sip:[local_ip]:[local_port]" />
+ <assignstr assign_to="rr_rcv" value="<test1-subsystem2@[local_ip]:[local_port];lr>,<test1-subsystem1@[local_ip]:[local_port];lr>" />
+ </action>
+ </nop>
+
+ <label id="4"/>
+
+ <recv request="INVITE" crlf="true" />
+
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Record-Route: [$rr_out]
+ Contact: <sip:[local_ip]:[local_port]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Record-Route: [$rr_out]
+ Contact: <sip:[local_ip]:[local_port]>
+ 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">
+ <action>
+ <ereg regexp="ACK\s*([^ ]*) SIP/2.0" search_in="msg" assign_to="1,uri_got"/>
+ <strcmp assign_to="1" variable="uri_got" variable2="rr_uri" />
+ <test assign_to="url_wrong" variable="1" compare="not_equal" value="" />
+
+ <ereg regexp="\s*(.*)\s*" search_in="hdr" header="Route:" assign_to="1,r_got"/>
+ <strcmp assign_to="1" variable="r_got" variable2="rr_rcv" />
+ <test assign_to="r_wrong" variable="1" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="url_wrong">
+ <action>
+ <error message="ACK URI expected '[$rr_uri]' but got '[$uri_got]'" />
+ </action>
+ </nop>
+ <nop condexec="r_wrong">
+ <action>
+ <error message="ACK Route expected '[$rr_rcv]' but got '[$r_got]'" />
+ </action>
+ </nop>
+
+ <recv request="BYE">
+ <action>
+ <ereg regexp="BYE\s*([^ ]*) SIP/2.0" search_in="msg" assign_to="1,uri_got"/>
+ <strcmp assign_to="1" variable="uri_got" variable2="rr_uri" />
+ <test assign_to="url_wrong" variable="1" compare="not_equal" value="" />
+
+ <ereg regexp="\s*(.*)\s*" search_in="hdr" header="Route:" assign_to="1,r_got"/>
+ <strcmp assign_to="1" variable="r_got" variable2="rr_rcv" />
+ <test assign_to="r_wrong" variable="1" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="url_wrong">
+ <action>
+ <error message="BYE URI expected '[$rr_uri]' but got '[$uri_got]'" />
+ </action>
+ </nop>
+ <nop condexec="r_wrong">
+ <action>
+ <error message="BYE Route expected '[$rr_rcv]' but got '[$r_got]'" />
+ </action>
+ </nop>
+
+ <send>
+ <![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>
+
+</scenario>
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svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/route/sipp/send-invite.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/route/sipp/send-invite.xml?view=auto&rev=4681
==============================================================================
--- asterisk/trunk/tests/channels/SIP/route/sipp/send-invite.xml (added)
+++ asterisk/trunk/tests/channels/SIP/route/sipp/send-invite.xml Mon Feb 10 12:25:08 2014
@@ -1,0 +1,176 @@
+<?xml version="1.0" encoding="iso-8859-2" ?>
+<!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 -->
+
+<scenario name="send-invite">
+ <nop hide="true">
+ <action>
+ <assignstr assign_to="1" value="[call_number]" />
+ <strcmp assign_to="rr_type" variable="1" value="1" />
+ </action>
+ </nop>
+
+ <nop hide="true" next="4" test="rr_type">
+ <action>
+ <assignstr assign_to="rr_out" value="<strict1@[local_ip]:[local_port]>,<strict2@[local_ip]:[local_port]>" />
+ <assignstr assign_to="rr_rcv" value="[$rr_out]" />
+ <assignstr assign_to="bye_uri" value="strict1@[local_ip]:[local_port]" />
+ <assignstr assign_to="rr_bye" value="<strict2@[local_ip]:[local_port]>,<sip:51@[local_ip]:[local_port]>" />
+ </action>
+ </nop>
+
+ <nop hide="true">
+ <action>
+ <assignstr assign_to="rr_out" value="<loose1@[local_ip]:[local_port];lr>,<loose2@[local_ip]:[local_port];lr>" />
+ <assignstr assign_to="rr_rcv" value="[$rr_out]" />
+ <assignstr assign_to="bye_uri" value="sip:51@[local_ip]:[local_port]" />
+ <assignstr assign_to="rr_bye" value="<loose1@[local_ip]:[local_port];lr>,<loose2@[local_ip]:[local_port];lr>" />
+ </action>
+ </nop>
+
+ <label id="4"/>
+
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:52@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:51@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:52@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: sip:51@[local_ip]:[local_port]
+ Record-Route: [$rr_out]
+ Max-Forwards: 100
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 8
+ a=rtpmap:8 PCMA/8000
+
+ ]]>
+ </send>
+
+ <recv response="100">
+ <action>
+ <ereg regexp="\s*(.*)\s*" search_in="hdr" header="Record-Route:" assign_to="1,rr_got"/>
+ <strcmp assign_to="1" variable="rr_got" variable2="rr_rcv" />
+ <test assign_to="r_wrong" variable="1" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="r_wrong">
+ <action>
+ <error message="100 RR expected '[$rr_rcv]' but got '[$rr_got]'" />
+ </action>
+ </nop>
+
+ <recv response="180">
+ <action>
+ <ereg regexp="\s*(.*)\s*" search_in="hdr" header="Record-Route:" assign_to="1,rr_got"/>
+ <strcmp assign_to="1" variable="rr_got" variable2="rr_rcv" />
+ <test assign_to="r_wrong" variable="1" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="r_wrong">
+ <action>
+ <error message="180 RR expected '[$rr_rcv]' but got '[$rr_got]'" />
+ </action>
+ </nop>
+
+ <recv response="183" optional="true">
+ <action>
+ <ereg regexp="\s*(.*)\s*" search_in="hdr" header="Record-Route:" assign_to="1,rr_got"/>
+ <strcmp assign_to="1" variable="rr_got" variable2="rr_rcv" />
+ <test assign_to="r_wrong" variable="1" compare="not_equal" value="" check_it="true" />
+ </action>
+ </recv>
+
+ <recv response="200" rrs="true">
+ <action>
+ <ereg regexp="\s*(.*)\s*" search_in="hdr" header="Record-Route:" assign_to="1,rr_got"/>
+ <strcmp assign_to="1" variable="rr_got" variable2="rr_rcv" />
+ <test assign_to="r_wrong" variable="1" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="r_wrong">
+ <action>
+ <error message="200 RR expected '[$rr_rcv]' but got '[$rr_got]'" />
+ </action>
+ </nop>
+
+ <send>
+ <![CDATA[
+ ACK [next_url] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:51@[local_ip]:[local_port]>;tag=[call_number]
+ [last_To:]
+ [routes]
+ Call-ID: [call_id]
+ CSeq: [cseq] ACK
+ Contact: sip:51@[local_ip]:[local_port]
+ Max-Forwards: 100
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE">
+ <action>
+ <ereg regexp="BYE\s*([^ ]*) SIP/2.0" search_in="msg" assign_to="1,bye_got"/>
+ <strcmp assign_to="1" variable="bye_got" variable2="bye_uri" />
+ <test assign_to="url_wrong" variable="1" compare="not_equal" value="" />
+
+ <ereg regexp="\s*(.*)\s*" search_in="hdr" header="Route:" assign_to="1,rr_got"/>
+ <strcmp assign_to="1" variable="rr_got" variable2="rr_bye" />
+ <test assign_to="r_wrong" variable="1" compare="not_equal" value="" />
+ </action>
+ </recv>
+
+ <nop condexec="url_wrong">
+ <action>
+ <error message="BYE URI expected '[$bye_uri]' but got '[$bye_got]'" />
+ </action>
+ </nop>
+ <nop condexec="r_wrong">
+ <action>
+ <error message="BYE Route expected '[$rr_bye]' but got '[$rr_got]'" />
+ </action>
+ </nop>
+
+ <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>
+
+</scenario>
+
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Added: asterisk/trunk/tests/channels/SIP/route/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/route/test-config.yaml?view=auto&rev=4681
==============================================================================
--- asterisk/trunk/tests/channels/SIP/route/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/route/test-config.yaml Mon Feb 10 12:25:08 2014
@@ -1,0 +1,41 @@
+testinfo:
+ summary: 'SIP Record-Route Support'
+ description: |
+ 'This test verifies SIP Record-Route handling.
+ Test includes loose and strict routes.'
+
+properties:
+ minversion: '1.8.0.0'
+ dependencies:
+ - python : 'twisted'
+ - python : 'starpy'
+ - app : 'sipp'
+ - asterisk : 'chan_sip'
+ tags:
+ - SIP
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ # Need to originate exactly 2 channels for recv-invite.
+ -
+ config-section: originator-recv-invite
+ typename: 'pluggable_modules.Originator'
+ -
+ config-section: originator-recv-invite
+ typename: 'pluggable_modules.Originator'
+
+originator-recv-invite:
+ application: 'Wait'
+ data: '1'
+ trigger: 'ami_connect'
+
+sipp-config:
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'send-invite.xml', '-p': '5061', '-m': '2', },}
+ - { 'key-args': {'scenario': 'recv-invite.xml', '-p': '5062', '-m': '2', },}
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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=4681&r1=4680&r2=4681
==============================================================================
--- asterisk/trunk/tests/channels/SIP/tests.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/tests.yaml Mon Feb 10 12:25:08 2014
@@ -60,6 +60,7 @@
- dir: 'session_timers'
- test: 'path'
- test: 'alwaysauthreject'
+ - test: 'route'
- dir: 'register_forbidden_retry'
- test: 'outbound_register_from'
- test: 'outbound_reregister_from'
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