[svn-commits] kmoore: testsuite/asterisk/trunk r2995 - in /asterisk/trunk/tests/channels/SI...

SVN commits to the Digium repositories svn-commits at lists.digium.com
Thu Jan 19 15:24:15 CST 2012


Author: kmoore
Date: Thu Jan 19 15:24:12 2012
New Revision: 2995

URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=2995
Log:
Add new test scenario and simplify addition of new tests

This scenario tests for proper declination behavior for encrypted video streams
when video support is disabled.  This patch also removes the requirement for
manual numbering of SIP ports for each scenario within the codec_negotiation
test.

Added:
    asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_crypto.xml   (with props)
Modified:
    asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/sip.conf
    asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test

Modified: asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/sip.conf?view=diff&rev=2995&r1=2994&r2=2995
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/sip.conf (original)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/sip.conf Thu Jan 19 15:24:12 2012
@@ -17,3 +17,10 @@
 allow=ulaw
 allow=h261
 allow=t140
+
+[guest2]
+type=user
+insecure=invite,port
+videosupport=no
+t38pt_udptl=no
+textsupport=no

Modified: asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test?view=diff&rev=2995&r1=2994&r2=2995
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test Thu Jan 19 15:24:12 2012
@@ -18,62 +18,33 @@
 TEST_DIR = os.path.dirname(os.path.realpath(__file__))
 
 SIPP_SCENARIOS = [
-    {
-        'scenario' : 'single_audio.xml',
-        '-p' : '5061'
-    },
-    {
-        'scenario' : 'decline_incompat_audio.xml',
-        '-p' : '5062'
-    },
-    {
-        'scenario' : 'single_video.xml',
-        '-p' : '5063'
-    },
-    {
-        'scenario' : 'single_video_inverse.xml',
-        '-p' : '5064'
-    },
-    {
-        'scenario' : 'decline_incompat_video.xml',
-        '-p' : '5065'
-    },
-    {
-        'scenario' : 'single_text.xml',
-        '-p' : '5066'
-    },
-    {
-        'scenario' : 'single_text_inverse.xml',
-        '-p' : '5067'
-    },
-    {
-        'scenario' : 'decline_incompat_text.xml',
-        '-p' : '5068'
-    },
-    {
-        'scenario' : 'single_image.xml',
-        '-p' : '5069'
-    },
-    {
-        'scenario' : 'single_image_inverse.xml',
-        '-p' : '5070'
-    },
-    {
-        'scenario' : 'avt_streams.xml',
-        '-p' : '5071'
-    },
-    {
-        'scenario' : 'multistream.xml',
-        '-p' : '5072'
-    },
-    {
-        'scenario' : 'orderstream.xml',
-        '-p' : '5073'
-    }
+    {'scenario' : 'single_audio.xml',},
+    {'scenario' : 'decline_incompat_audio.xml',},
+    {'scenario' : 'single_video.xml',},
+    {'scenario' : 'single_video_inverse.xml',},
+    {'scenario' : 'decline_incompat_video.xml',},
+    {'scenario' : 'single_text.xml',},
+    {'scenario' : 'single_text_inverse.xml',},
+    {'scenario' : 'decline_incompat_text.xml',},
+    {'scenario' : 'single_image.xml',},
+    {'scenario' : 'single_image_inverse.xml',},
+    {'scenario' : 'avt_streams.xml',},
+    {'scenario' : 'multistream.xml',},
+    {'scenario' : 'fax_sim.xml',},
+    {'scenario' : 'decline_crypto.xml',},
+    {'scenario' : 'orderstream.xml',},
 ]
+
+# set port numberings and timeouts
+port = 5061
+def update_entry(entry):
+    global port
+    entry['-p'] = "%d" % port
+    port += 1
 
 # generate SIPP scenarios with appropriate port numbers and the config to go with it
 def main():
+    [update_entry(i) for i in SIPP_SCENARIOS]
     test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
     return test.run()
 

Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_crypto.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_crypto.xml?view=auto&rev=2995
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_crypto.xml (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_crypto.xml Thu Jan 19 15:24:12 2012
@@ -1,0 +1,54 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:guest2@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: test1 <sip:guest2@[local_ip]:[local_port]>;tag=[call_number]
+      To: test <sip:guest2@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:guest2@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      User-Agent: Channel Param Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=guest1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=video 6002 RTP/SAVP 32 34
+      a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
+      a=rtpmap:32 MPV/90000
+      a=rtpmap:34 H263/90000
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <recv response="488" rtd="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"/>
+
+</scenario>
+

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