[asterisk-commits] file: testsuite/asterisk/trunk r6303 - in /asterisk/trunk/tests/channels/pjsi...

SVN commits to the Asterisk project asterisk-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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