[svn-commits] file: testsuite/asterisk/trunk r6303 - in /asterisk/trunk/tests/channels/pjsi...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jan 19 07:19:22 CST 2015
Author: file
Date: Mon Jan 19 07:19:19 2015
New Revision: 6303
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=6303
Log:
res_pjsip / res_pjsip_multihomed: Add test for checking of Contact on UAS session.
This test confirms that the Contact received in the 200 OK when sending a call
into PJSIP contains the correct address information when PJSIP is configured with
multiple transports.
ASTERISK-24615
Reported by: David Justl
Review: https://reviewboard.asterisk.org/r/4335/
Added:
asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/
asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/
asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/
asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/sipp/
asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/sipp/answer.xml (with props)
asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/pjsip/tests.yaml
Added: asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/extensions.conf?view=auto&rev=6303
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/extensions.conf Mon Jan 19 07:19:19 2015
@@ -1,0 +1,4 @@
+[default]
+exten => echo,1,Answer()
+same => n,Echo()
+same => n,Hangup()
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Added: asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/pjsip.conf?view=auto&rev=6303
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/configs/ast1/pjsip.conf Mon Jan 19 07:19:19 2015
@@ -1,0 +1,20 @@
+[local-transport-1]
+type=transport
+bind=127.0.0.1
+protocol=udp
+
+[local-transport-2]
+type=transport
+bind=127.0.0.1:5070
+protocol=udp
+
+[local-transport-3]
+type=transport
+bind=127.0.0.1:5080
+protocol=udp
+
+[alice]
+type=endpoint
+context=default
+allow=!all,ulaw,alaw
+media_address=127.0.0.1
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Added: asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/sipp/answer.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/sipp/answer.xml?view=auto&rev=6303
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/sipp/answer.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/sipp/answer.xml Mon Jan 19 07:19:19 2015
@@ -1,0 +1,95 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in initial INVITE">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <!-- Check the Contact header. -->
+ <ereg regexp="sip:127.0.0.1:5070"
+ header="Contact"
+ search_in="hdr"
+ check_it="true"
+ assign_to="contact"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:echo@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:test@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: 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"/>
+
+ <Reference variables="contact" />
+
+</scenario>
+
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Added: asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/test-config.yaml?view=auto&rev=6303
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/incoming_call_on_second_transport/test-config.yaml Mon Jan 19 07:19:19 2015
@@ -1,0 +1,29 @@
+testinfo:
+ summary: 'Tests an incoming call on a second configured transport'
+ description: |
+ 'Run a SIPp scenario that sends a call into chan_pjsip. Confirms that the 200 OK received back
+ contains the port that the request was sent to.'
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+
+test-object-config:
+ fail-on-any: False
+ test-iterations:
+ -
+ scenarios:
+ - { 'key-args': {'scenario': 'answer.xml', '-i': '127.0.0.1', '-p': '5061', '-d': '5000', '-s': 'alice', '-rsa': '127.0.0.1:5070'} }
+
+properties:
+ minversion: '13.3.0'
+ dependencies:
+ - sipp :
+ version : 'v3.3'
+ - asterisk : 'res_pjsip'
+ - asterisk : 'chan_pjsip'
+ - asterisk : 'res_pjsip_session'
+ - asterisk : 'res_pjsip_multihomed'
+ tags:
+ - pjsip
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Modified: asterisk/trunk/tests/channels/pjsip/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/tests.yaml?view=diff&rev=6303&r1=6302&r2=6303
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/tests.yaml (original)
+++ asterisk/trunk/tests/channels/pjsip/tests.yaml Mon Jan 19 07:19:19 2015
@@ -2,6 +2,7 @@
tests:
- test: 'handle_options_request'
- test: 'incoming_calls_without_auth'
+ - test: 'incoming_call_on_second_transport'
- dir: 'basic_calls'
- dir: 'sdp_offer_answer'
- test: 'srtp_negotiation'
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