[asterisk-commits] mjordan: testsuite/asterisk/trunk r5289 - in /asterisk/trunk/tests/channels/S...

SVN commits to the Asterisk project asterisk-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>
+

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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>
+

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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>

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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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