[Asterisk-code-review] H.263+: Remove attribute pass-through tests. (testsuite[master])
Alexander Traud
asteriskteam at digium.com
Tue Feb 23 04:41:34 CST 2021
Alexander Traud has uploaded this change for review. ( https://gerrit.asterisk.org/c/testsuite/+/15491 )
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(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/91/15491/1
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
Gerrit-Branch: master
Gerrit-Change-Id: Ia45e3f4a003723951c5963dd6a77885447749474
Gerrit-Change-Number: 15491
Gerrit-PatchSet: 1
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-MessageType: newchange
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