[asterisk-commits] kmoore: testsuite/asterisk/trunk r2678 - in /asterisk/trunk/tests/channels/SI...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 7 11:12:58 CST 2011
Author: kmoore
Date: Mon Nov 7 11:12:51 2011
New Revision: 2678
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=2678
Log:
Testsuite: Add tests for basic SDP codec and stream negotiation in SIP
Add tests for situations involving single stream, incompatible stream, and
multistream situations. This test is skipped until several fixes are finished
with testing and committed.
(closes issue ASTERISK-18445)
Review: https://reviewboard.asterisk.org/r/1520/
Added:
asterisk/trunk/tests/channels/SIP/codec_negotiation/
asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/
asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/
asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/avt_streams.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_audio.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_text.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_video.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/multistream.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/orderstream.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_audio.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_image.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_image_inverse.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_text.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_text_inverse.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_video.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_video_inverse.xml (with props)
asterisk/trunk/tests/channels/SIP/codec_negotiation/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/tests.yaml
Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/extensions.conf?view=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/extensions.conf Mon Nov 7 11:12:51 2011
@@ -1,0 +1,6 @@
+[default]
+
+exten => _.,1,NoOp()
+
+exten => test,1,Answer()
+exten => test,n,Wait(30) ; SIPp will hang up.
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Added: 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=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/configs/ast1/sip.conf Mon Nov 7 11:12:51 2011
@@ -1,0 +1,19 @@
+[general]
+
+textsupport=yes
+videosupport=yes
+t38pt_udptl=yes
+allowguest=no
+
+[guest0]
+type=user
+insecure=invite,port
+t38pt_udptl=yes
+
+[guest1]
+type=user
+insecure=invite,port
+disallow=all
+allow=ulaw
+allow=h261
+allow=t140
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Added: 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=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/run-test Mon Nov 7 11:12:51 2011
@@ -1,0 +1,85 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2011, Digium, Inc.
+Kinsey Moore <kmoore at digium.com>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+'''
+
+import sys
+import os
+
+sys.path.append("lib/python")
+from asterisk.sipp import SIPpTest
+
+
+WORKING_DIR = "sip/codec_negotiation"
+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'
+ }
+]
+
+# generate SIPP scenarios with appropriate port numbers and the config to go with it
+def main():
+ test = SIPpTest(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+ return test.run()
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
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Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/avt_streams.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/avt_streams.xml?view=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/avt_streams.xml (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/avt_streams.xml Mon Nov 7 11:12:51 2011
@@ -1,0 +1,140 @@
+<?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=text 40000 RTP/AVP 106
+ a=rtpmap:106 T140/1000
+
+ ]]>
+ </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 and 2 video streams, even if one or more streams are declined -->
+ <ereg regexp="m=audio [0-9]{1,5} RTP/AVP .*m=video [0-9]{1,5} RTP/AVP .*m=text [0-9]{1,5} 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>
+
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Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_audio.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_audio.xml?view=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_audio.xml (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_audio.xml Mon Nov 7 11:12:51 2011
@@ -1,0 +1,52 @@
+<?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:guest1@[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: 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=audio 6000 RTP/AVP 8
+ a=rtpmap:8 PCMA/8000
+
+ ]]>
+ </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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Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_text.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_text.xml?view=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_text.xml (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_text.xml Mon Nov 7 11:12:51 2011
@@ -1,0 +1,52 @@
+<?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:guest1@[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: 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=text 40000 RTP/AVP 100
+ a=rtpmap:100 red/1000
+
+ ]]>
+ </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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Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_video.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_video.xml?view=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_video.xml (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/decline_incompat_video.xml Mon Nov 7 11:12:51 2011
@@ -1,0 +1,53 @@
+<?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:guest1@[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: 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/AVP 32 34
+ 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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Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/multistream.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/multistream.xml?view=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/multistream.xml (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/multistream.xml Mon Nov 7 11:12:51 2011
@@ -1,0 +1,155 @@
+<?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 113
+ a=rtpmap:113 H264/90000
+ a=fmtp:113 profile-level-id=ABCDEF;packetization-mode=1
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="183" optional="true">
+ </recv>
+
+ <recv response="200" rtd="true" optional="true" next="1">
+ <action>
+ <!-- ensure that we get back 1 audio and 2 video streams, even if one or more streams are declined -->
+ <ereg regexp="m=audio [0-9]{1,5} RTP/AVP .*m=video [0-9]{1,5} RTP/AVP .*m=video [0-9]{1,5} RTP/AVP "
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <recv response="488" rtd="true" optional="true" next="2">
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <!-- fail a bad 200 -->
+ <exec int_cmd="stop_call"/>
+ </action>
+ </recv>
+
+ <label id="1"/>
+ <!-- successful call flow -->
+ <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>
+
+ <label id="2"/>
+
+ <!-- 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/codec_negotiation/sipp/multistream.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/multistream.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/multistream.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/orderstream.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/orderstream.xml?view=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/orderstream.xml (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/orderstream.xml Mon Nov 7 11:12:51 2011
@@ -1,0 +1,154 @@
+<?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=text 40000 RTP/AVP 106
+ a=rtpmap:106 T140/1000
+ m=video 6002 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" optional="true" next="1">
+ <action>
+ <!-- ensure that we get back 1 audio, 1 text, and 1 video streams in that order, even if one or more streams are declined -->
+ <ereg regexp="m=audio [0-9]{1,5} RTP/AVP .*m=text [0-9]{1,5} RTP/AVP .*m=video [0-9]{1,5} RTP/AVP "
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <recv response="488" rtd="true" optional="true" next="2">
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <!-- fail a bad 200 -->
+ <exec int_cmd="stop_call"/>
+ </action>
+ </recv>
+
+ <label id="1"/>
+ <!-- successful call flow -->
+ <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>
+
+ <label id="2"/>
+
+ <!-- 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/codec_negotiation/sipp/orderstream.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/orderstream.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/orderstream.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_audio.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_audio.xml?view=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_audio.xml (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_audio.xml Mon Nov 7 11:12:51 2011
@@ -1,0 +1,133 @@
+<?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:guest1@[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: 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=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
+
+ ]]>
+ </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>
+ <ereg regexp="m=audio [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..?a=rtpmap:0 PCMU/8000"
+ 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:guest1@[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:guest1@[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/codec_negotiation/sipp/single_audio.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_audio.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_audio.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_image.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_image.xml?view=auto&rev=2678
==============================================================================
--- asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_image.xml (added)
+++ asterisk/trunk/tests/channels/SIP/codec_negotiation/sipp/single_image.xml Mon Nov 7 11:12:51 2011
@@ -1,0 +1,94 @@
+<?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:guest1@[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: 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=image 40000 udptl t38
+ a=T38FaxUdpEC:t38UDPRedundancy
+
+ ]]>
+ </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>
+ <ereg regexp="m=image [0-9]{1,5} udptl t38"
+ 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:guest1@[local_ip]:[local_port]>;tag=[call_number]
[... 662 lines stripped ...]
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