[svn-commits] kmoore: testsuite/asterisk/trunk r3313 - in /asterisk/trunk/tests/channels/SI...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jul 12 14:32:03 CDT 2012
Author: kmoore
Date: Thu Jul 12 14:31:59 2012
New Revision: 3313
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=3313
Log:
Add new test for SIP stream declination and clean up test
This adds a new test to the SDP_offer_answer set of SIPp-based tests
for handling multiple streams and cleans up the usage of SIPpTest now
that port numbers are set automatically.
Added:
asterisk/trunk/tests/channels/SIP/SDP_offer_answer/sipp/decline_multistream.xml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/SDP_offer_answer/run-test
Modified: asterisk/trunk/tests/channels/SIP/SDP_offer_answer/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_offer_answer/run-test?view=diff&rev=3313&r1=3312&r2=3313
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_offer_answer/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/SDP_offer_answer/run-test Thu Jul 12 14:31:59 2012
@@ -14,6 +14,7 @@
from twisted.internet import reactor
from asterisk.sipp import SIPpTest
+from asterisk.version import AsteriskVersion
WORKING_DIR = "SIP/SDP_offer_answer"
@@ -37,16 +38,10 @@
{'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]
+ if AsteriskVersion() > AsteriskVersion("11"):
+ SIPP_SCENARIOS.append({'scenario' : 'decline_multistream.xml'})
test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
reactor.run()
if not test.passed:
Added: asterisk/trunk/tests/channels/SIP/SDP_offer_answer/sipp/decline_multistream.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_offer_answer/sipp/decline_multistream.xml?view=auto&rev=3313
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_offer_answer/sipp/decline_multistream.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_offer_answer/sipp/decline_multistream.xml Thu Jul 12 14:31:59 2012
@@ -1,0 +1,142 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:guest0@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:kartoffelsalat@[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=user1 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 3 4 5 7 8 9 10 18 96 97 100 101 102 107 108 110 111 112 115 116 117
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:3 GSM/8000
+ a=rtpmap:4 G723/8000
+ a=fmtp:4 annexa=no
+ a=rtpmap:5 DVI4/8000
+ a=rtpmap:7 LPC/8000
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:9 G722/8000
+ a=rtpmap:10 L16/8000
+ a=rtpmap:18 G729/8000
+ a=fmtp:18 annexb=no
+ a=rtpmap:96 SILK/8000
+ a=fmtp:96 maxaveragebitrate=10000
+ a=fmtp:96 usedtx=0
+ a=fmtp:96 useinbandfec=1
+ a=rtpmap:97 iLBC/8000
+ a=fmtp:97 mode=30
+ a=rtpmap:100 SILK/12000
+ a=fmtp:100 maxaveragebitrate=12000
+ a=fmtp:100 usedtx=0
+ a=fmtp:100 useinbandfec=1
+ a=rtpmap:101 telephone-event/8000
+ a=rtpmap:102 G7221/16000
+ a=fmtp:102 bitrate=32000
+ a=rtpmap:107 SILK/16000
+ a=fmtp:107 maxaveragebitrate=20000
+ a=fmtp:107 usedtx=0
+ a=fmtp:107 useinbandfec=1
+ a=rtpmap:108 SILK/24000
+ a=fmtp:108 maxaveragebitrate=30000
+ a=fmtp:108 usedtx=0
+ a=fmtp:108 useinbandfec=1
+ a=rtpmap:110 speex/8000
+ a=rtpmap:111 G726-32/8000
+ a=rtpmap:112 AAL2-G726-32/8000
+ a=rtpmap:115 G7221/32000
+ a=fmtp:115 bitrate=48000
+ a=rtpmap:116 G719/48000
+ a=fmtp:116 bitrate=64000
+ a=rtpmap:117 speex/16000
+ m=video 6002 RTP/AVP 31 32 34
+ a=rtpmap:31 H261/90000
+ a=rtpmap:32 MPV/90000
+ a=rtpmap:34 H263/90000
+ m=video 6003 RTP/AVP 31 32 34
+ a=rtpmap:31 H261/90000
+ 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="200" rtd="true">
+ <action>
+ <!-- ensure that we get back 1 audio stream, 1 declined video stream and 1 video stream -->
+ <ereg regexp="m=audio [0-9]{1,5} RTP/AVP .*m=video [0-9]{1,5} RTP/AVP .*m=video 0 RTP/AVP "
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:guest0@[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:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:guest0@[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:kartoffelsalat@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance 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"/>
+
+</scenario>
+
Propchange: asterisk/trunk/tests/channels/SIP/SDP_offer_answer/sipp/decline_multistream.xml
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