[svn-commits] mjordan: testsuite/asterisk/trunk r5289 - in /asterisk/trunk/tests/channels/S...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jul 21 20:44:19 CDT 2014
Author: mjordan
Date: Mon Jul 21 20:44:12 2014
New Revision: 5289
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=5289
Log:
channels/SIP/SDP_attribute_passthrough: Update tests for Asterisk 13
This patch updates the SDP_attribute_passthrough test for the expected ordering
of attributes in Asterisk 13. The attributes are generally idential to what
would be offered/answered previously, but the ordering has changed.
Added:
asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml (with props)
asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264_13.xml (with props)
asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml (with props)
asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test
Modified: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test?view=diff&rev=5289&r1=5288&r2=5289
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test (original)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/run-test Mon Jul 21 20:44:12 2014
@@ -15,6 +15,7 @@
sys.path.append("lib/python")
from asterisk.asterisk import Asterisk
+from asterisk.version import AsteriskVersion
from asterisk.test_case import TestCase
from asterisk.sipp import SIPpScenario
from twisted.internet import reactor
@@ -26,12 +27,19 @@
def __init__(self):
TestCase.__init__(self)
self.create_asterisk()
- self.sipp_phone_a_scenarios = [{'scenario':'phone_A_h263.xml','-i':'127.0.0.2','-p':'5061'},
- {'scenario':'phone_A_h264.xml','-i':'127.0.0.2','-p':'5062'},
- {'scenario':'phone_A_speex.xml','-i':'127.0.0.2','-p':'5065'}]
- self.sipp_phone_b_scenarios = [{'scenario':'phone_B_h263.xml','-i':'127.0.0.3','-p':'5063'},
- {'scenario':'phone_B_h264.xml','-i':'127.0.0.3','-p':'5064'},
- {'scenario':'phone_B_speex.xml','-i':'127.0.0.3','-p':'5066'}]
+ self.sipp_phone_a_scenarios = [{'scenario':'phone_A_speex.xml','-i':'127.0.0.2','-p':'5065'},]
+ self.sipp_phone_b_scenarios = [{'scenario':'phone_B_speex.xml','-i':'127.0.0.3','-p':'5066'},]
+ if (AsteriskVersion() < AsteriskVersion("13")):
+ self.sipp_phone_a_scenarios.extend([{'scenario':'phone_A_h263.xml','-i':'127.0.0.2','-p':'5061'},
+ {'scenario':'phone_A_h264.xml','-i':'127.0.0.2','-p':'5062'},])
+
+ self.sipp_phone_b_scenarios.extend([{'scenario':'phone_B_h263.xml','-i':'127.0.0.3','-p':'5063'},
+ {'scenario':'phone_B_h264.xml','-i':'127.0.0.3','-p':'5064'},])
+ else:
+ self.sipp_phone_a_scenarios.extend([{'scenario':'phone_A_h263_13.xml','-i':'127.0.0.2','-p':'5061'},
+ {'scenario':'phone_A_h264_13.xml','-i':'127.0.0.2','-p':'5062'},])
+ self.sipp_phone_b_scenarios.extend([{'scenario':'phone_B_h263_13.xml','-i':'127.0.0.3','-p':'5063'},
+ {'scenario':'phone_B_h264_13.xml','-i':'127.0.0.3','-p':'5064'},])
self.passed = True
self.__test_counter = 0
Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml?view=auto&rev=5289
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml Mon Jul 21 20:44:12 2014
@@ -1,0 +1,97 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test-h263@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[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=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
+ m=video 6002 RTP/AVP 34
+ a=rtpmap:34 H263/90000
+ a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;CIF16=1;F=1;I=1;J=1;T=1;K=1;N=1;PAR=255:255;BPP=65535;HRD=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">
+ <action>
+ <ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:34 H263/90000.*a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;CIF16=1;VGA=0;F=1;I=1;J=1;T=1;K=1;N=1;BPP=65535;HRD=1;PAR=255:255"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test-h263@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[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-h263@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[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_attribute_passthrough/sipp/phone_A_h263_13.xml
------------------------------------------------------------------------------
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------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264_13.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264_13.xml?view=auto&rev=5289
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264_13.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264_13.xml Mon Jul 21 20:44:12 2014
@@ -1,0 +1,97 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Channel Test">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test-h264@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[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=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
+ m=video 6002 RTP/AVP 99
+ a=rtpmap:99 H264/90000
+ a=fmtp:99 profile-level-id=42801e;packetization-mode=1;max-mbps=48600
+
+ ]]>
+ </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=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:99 H264/90000.*a=fmtp:99 max-mbps=48600;packetization-mode=1;profile-level-id=42801E"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test-h264@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[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-h264@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test1 <sip:phoneA@[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_attribute_passthrough/sipp/phone_A_h264_13.xml
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264_13.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h264_13.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml?view=auto&rev=5289
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml Mon Jul 21 20:44:12 2014
@@ -1,0 +1,89 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B INVITE with H.263 and answer with H.263">
+ <Global variables="global_call_id"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ <ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:34 H263/90000.*a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;CIF16=1;VGA=0;F=1;I=1;J=1;T=1;K=1;N=1;BPP=65535;HRD=1;PAR=255:255"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <pause milliseconds="200"/>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=guest3 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 34
+ a=rtpmap:34 H263/90000
+ a=fmtp:34 SQCIF=1;QCIF=1;CIF=1;CIF4=1;CIF16=1;F=1;I=1;J=1;T=1;K=1;N=1;PAR=255:255;BPP=65535;HRD=1
+
+ ]]>
+ </send>
+
+ <!-- RECV ACK -->
+ <recv request="ACK"/>
+
+ <recv request="BYE"/>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+</scenario>
Propchange: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml
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svn:eol-style = native
Propchange: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml?view=auto&rev=5289
==============================================================================
--- asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml (added)
+++ asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml Mon Jul 21 20:44:12 2014
@@ -1,0 +1,89 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Phone B INVITE with H.264 and answer with H.264">
+ <Global variables="global_call_id"/>
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*"
+ header="Call-ID:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="global_call_id"/>
+ <ereg regexp="m=video [0-9]{1,5} RTP/AVP( [0-9]{1,3})+..*a=rtpmap:99 H264/90000.*a=fmtp:99 max-mbps=48600;packetization-mode=1;profile-level-id=42801E"
+ search_in="body" check_it="true" assign_to="1"/>
+ <strcmp assign_to="1" variable="1" value=""/>
+
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <pause milliseconds="200"/>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=guest3 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 99
+ a=rtpmap:99 H264/90000
+ a=fmtp:99 profile-level-id=42801e;packetization-mode=1;max-mbps=48600
+
+ ]]>
+ </send>
+
+ <!-- RECV ACK -->
+ <recv request="ACK"/>
+
+ <recv request="BYE"/>
+
+ <send retrans="500">
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:bob@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
+ Supported: 100rel,replaces
+ User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
+ Accept-Language: en
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+</scenario>
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svn:keywords = Author Date Id Revision
Propchange: asterisk/trunk/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h264_13.xml
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