[Asterisk-code-review] H.263+: Remove attribute pass-through tests. (testsuite[16])

Friendly Automation asteriskteam at digium.com
Wed Mar 3 09:22:04 CST 2021


Friendly Automation has submitted this change. ( https://gerrit.asterisk.org/c/testsuite/+/15508 )

Change subject: H.263+: Remove attribute pass-through tests.
......................................................................

H.263+: Remove attribute pass-through tests.

General attribute pass-through is tested via H.264 already.
Therefore, testing H.263+ again would just be a duplicate test.
Finally, the order of the attributes changed in the H.263+ module.

Change-Id: Ia45e3f4a003723951c5963dd6a77885447749474
---
M tests/channels/SIP/SDP_attribute_passthrough/run-test
D tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml
D tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml
M tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/extensions.conf
M tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf
M tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/run-test
D tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_A_h263.xml
D tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h263.xml
8 files changed, 2 insertions(+), 392 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved
  Friendly Automation: Approved for Submit



diff --git a/tests/channels/SIP/SDP_attribute_passthrough/run-test b/tests/channels/SIP/SDP_attribute_passthrough/run-test
index dfffe2b..5d98f0a 100755
--- a/tests/channels/SIP/SDP_attribute_passthrough/run-test
+++ b/tests/channels/SIP/SDP_attribute_passthrough/run-test
@@ -27,10 +27,8 @@
         self.create_asterisk()
         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'},]
-        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.sipp_phone_a_scenarios.extend([{'scenario':'phone_A_h264_13.xml','-i':'127.0.0.2','-p':'5062'},])
+        self.sipp_phone_b_scenarios.extend([{'scenario':'phone_B_h264_13.xml','-i':'127.0.0.3','-p':'5064'},])
 
         self.passed = True
         self.__test_counter = 0
diff --git a/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml b/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml
deleted file mode 100644
index a8c0521..0000000
--- a/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_A_h263_13.xml
+++ /dev/null
@@ -1,97 +0,0 @@
-<?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>
-
diff --git a/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml b/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml
deleted file mode 100644
index a132490..0000000
--- a/tests/channels/SIP/SDP_attribute_passthrough/sipp/phone_B_h263_13.xml
+++ /dev/null
@@ -1,89 +0,0 @@
-<?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>
diff --git a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/extensions.conf b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/extensions.conf
index 7c5e76c..6f2995f 100644
--- a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/extensions.conf
+++ b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/extensions.conf
@@ -1,4 +1,3 @@
 [default]
-exten => test-h263,1,Dial(PJSIP/phoneB-h263)
 exten => test-h264,1,Dial(PJSIP/phoneB-h264)
 exten => test-speex,1,Dial(PJSIP/phoneB-speex)
diff --git a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf
index 5598d0d..9946d70 100644
--- a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf
+++ b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/configs/ast1/pjsip.conf
@@ -21,20 +21,9 @@
 [phoneA](endpoint-template)
 disallow=all
 allow=ulaw
-allow=h263
 allow=h264
 allow=speex
 
-[phoneB-h263](endpoint-template)
-aors=phoneB-h263
-disallow=all
-allow=ulaw
-allow=h263
-
-[phoneB-h263]
-type=aor
-contact=sip:127.0.0.3:5063
-
 [phoneB-h264](endpoint-template)
 aors=phoneB-h264
 disallow=all
diff --git a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/run-test b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/run-test
index 16f0e37..05ddf4d 100755
--- a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/run-test
+++ b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/run-test
@@ -26,10 +26,8 @@
         TestCase.__init__(self)
         self.create_asterisk(test_config={'memcheck-delay-stop': 7})
         self.sipp_phone_a_scenarios = [{'scenario':'phone_A_speex.xml','-i':'127.0.0.2','-p':'5065'},
-					{'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 = [{'scenario':'phone_B_speex.xml','-i':'127.0.0.3','-p':'5066'},
-					{'scenario':'phone_B_h263.xml','-i':'127.0.0.3','-p':'5063'},
 					{'scenario':'phone_B_h264.xml','-i':'127.0.0.3','-p':'5064'},]
 
         self.passed = True
diff --git a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_A_h263.xml b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_A_h263.xml
deleted file mode 100644
index a8c0521..0000000
--- a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_A_h263.xml
+++ /dev/null
@@ -1,97 +0,0 @@
-<?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>
-
diff --git a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h263.xml b/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h263.xml
deleted file mode 100644
index 311d1f7..0000000
--- a/tests/channels/pjsip/sdp_offer_answer/attribute_passthrough/speex_h263_h264/sipp/phone_B_h263.xml
+++ /dev/null
@@ -1,91 +0,0 @@
-<?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=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 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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Gerrit-Project: testsuite
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Gerrit-Change-Id: Ia45e3f4a003723951c5963dd6a77885447749474
Gerrit-Change-Number: 15508
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