[asterisk-commits] testsuite: Cleanup tests/fax/pjsip/directmedia reinvite t38 (testsuite[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Feb 24 16:07:42 CST 2016
Anonymous Coward #1000019 has submitted this change and it was merged.
Change subject: testsuite: Cleanup tests/fax/pjsip/directmedia_reinvite_t38
......................................................................
testsuite: Cleanup tests/fax/pjsip/directmedia_reinvite_t38
* Fix duplicate To tags on party A and B responses.
* Make A and B party contacts specify the transport.
* Add message tracking headers to quickly locate the messages in the debug
log.
* Removed unnecessary dialplan extensions.
* Added missing test dependencies to the test yaml file.
Change-Id: Id9e1fac0c8d4d77393f9eca953addabf8c38fcc8
---
M tests/fax/pjsip/directmedia_reinvite_t38/configs/ast1/extensions.conf
M tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_A.xml
M tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_B.xml
M tests/fax/pjsip/directmedia_reinvite_t38/test-config.yaml
4 files changed, 43 insertions(+), 42 deletions(-)
Approvals:
Mark Michelson: Looks good to me, approved
Anonymous Coward #1000019: Verified
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/tests/fax/pjsip/directmedia_reinvite_t38/configs/ast1/extensions.conf b/tests/fax/pjsip/directmedia_reinvite_t38/configs/ast1/extensions.conf
index a83cef4..f0d6b49 100644
--- a/tests/fax/pjsip/directmedia_reinvite_t38/configs/ast1/extensions.conf
+++ b/tests/fax/pjsip/directmedia_reinvite_t38/configs/ast1/extensions.conf
@@ -1,13 +1,4 @@
-[general]
-PHONE_TO_DIAL=PJSIP/endpoint_B/sip:127.0.0.3
-
[default]
-exten => bypassbridge,1,NoOp()
- same => n,Dial(PJSIP/endpoint_B/sip:127.0.0.3,,g)
- same => n,UserEvent(TestStatus, extension: bypassbridge)
- same => n,Hangup()
-
-; Dial with no options; use bridge set up based on peer definitions
exten => basicdial,1,NoOp()
same => n,Dial(PJSIP/endpoint_B/sip:127.0.0.3,,g)
same => n,UserEvent(TestStatus, extension: basicdial)
diff --git a/tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_A.xml b/tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_A.xml
index 2d378b4..796e322 100644
--- a/tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_A.xml
+++ b/tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_A.xml
@@ -1,21 +1,22 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
-<scenario name="Phone A Hold with IP and Media Restrictions">
+<scenario name="Phone A direct media receiving T.38 fax">
<!-- Initial invite - Call phone B -->
<send retrans="500">
<![CDATA[
INVITE sip:[field2]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
- From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
+ From: "[field0]" <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[field2]@[remote_ip]:[remote_port];user=phone>
CSeq: 1 INVITE
Call-ID: [call_id]
- Contact: <sip:[field0]@[local_ip]:[local_port]>
+ Contact: <sip:[field0]@[local_ip]:[local_port];transport=[transport]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
+ X-Testsuite-Track-Phone-A: 1
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
@@ -44,16 +45,17 @@
<send>
<![CDATA[
- ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
+ ACK sip:[field2]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
- From: [field0] <sip:[field0]@[remote_ip]>;tag=[call_number]
- To: <sip:[field1]@[remote_ip];user=phone>[peer_tag_param]
+ From: "[field0]" <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:[field2]@[remote_ip]:[remote_port];user=phone>[peer_tag_param]
CSeq: 1 ACK
Call-ID: [call_id]
- Contact: <sip:[field0]@[local_ip]:[local_port]>
+ Contact: <sip:[field0]@[local_ip]:[local_port];transport=[transport]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
+ X-Testsuite-Track-Phone-A: 2
Max-Forwards: 70
Content-Length: 0
]]>
@@ -80,7 +82,7 @@
SIP/2.0 200 OK
[last_Via:]
[last_From:]
- [last_To:];tag=[call_number]
+ [last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[field0]@[local_ip]:[local_port];transport=[transport]>
@@ -88,7 +90,7 @@
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
- Testsuite-Track-Phone-A: 1
+ X-Testsuite-Track-Phone-A: 3
Content-Type: application/sdp
Content-Length: [len]
@@ -115,7 +117,7 @@
SIP/2.0 200 OK
[last_Via:]
[last_From:]
- [last_To:];tag=[call_number]
+ [last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[field0]@[local_ip]:[local_port];transport=[transport]>
@@ -123,7 +125,7 @@
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
- Testsuite-Track-Phone-A: 2
+ X-Testsuite-Track-Phone-A: 4
Content-Type: application/sdp
Content-Length: [len]
@@ -153,7 +155,7 @@
SIP/2.0 200 OK
[last_Via:]
[last_From:]
- [last_To:];tag=[call_number]
+ [last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[field0]@[local_ip]:[local_port];transport=[transport]>
@@ -161,7 +163,7 @@
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
- Testsuite-Track-Phone-A: 3
+ X-Testsuite-Track-Phone-A: 5
Content-Type: application/sdp
Content-Length: [len]
@@ -187,16 +189,14 @@
SIP/2.0 200 OK
[last_Via:]
[last_From:]
- [last_To:];tag=[call_number]
+ [last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[field0]@[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
- Testsuite-Track-Phone-A: 5
- Content-Type: application/sdp
+ X-Testsuite-Track-Phone-A: 6
Content-Length: 0
]]>
</send>
diff --git a/tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_B.xml b/tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_B.xml
index e34edb0..4781ba0 100644
--- a/tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_B.xml
+++ b/tests/fax/pjsip/directmedia_reinvite_t38/sipp/endpoint_B.xml
@@ -1,7 +1,7 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
-<scenario name="Phone B Hold with Media Restrictions">
+<scenario name="Phone B direct media sending T.38 fax">
<Global variables="remote_tag"/>
<recv request="INVITE" crlf="true">
@@ -19,12 +19,13 @@
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
- [last_To:];tag=[call_number]
+ [last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
+ X-Testsuite-Track-Phone-B: 1
Content-Length: 0
]]>
</send>
@@ -41,6 +42,7 @@
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Allow-Events: talk,hold,conference
Accept-Language: en
+ X-Testsuite-Track-Phone-B: 2
Content-Length: 0
]]>
</send>
@@ -52,7 +54,7 @@
SIP/2.0 200 OK
[last_Via:]
[last_From:]
- [last_To:]
+ [last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
@@ -60,7 +62,7 @@
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
- Testsuite-Track-Phone-B-Media-Restrict: 1
+ X-Testsuite-Track-Phone-B: 3
Content-Type: application/sdp
Content-Length: [len]
@@ -100,7 +102,7 @@
SIP/2.0 200 OK
[last_Via:]
[last_From:]
- [last_To:]
+ [last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
@@ -108,7 +110,7 @@
Supported: 100rel,replaces
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
- Testsuite-Track-Phone-B-Media-Restrict: 2
+ X-Testsuite-Track-Phone-B: 4
Content-Type: application/sdp
Content-Length: [len]
@@ -135,14 +137,15 @@
<![CDATA[
INVITE sip:endpoint_B@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
- From: <sip:127.0.0.3>
+ From: <sip:127.0.0.3>;tag=[call_number]
To: [$remote_tag]
CSeq: [cseq] INVITE
[last_Call-ID:]
- Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
+ X-Testsuite-Track-Phone-B: 5
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
@@ -173,14 +176,15 @@
<![CDATA[
ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
- From: <sip:127.0.0.3>
+ From: <sip:127.0.0.3>;tag=[call_number]
To: [$remote_tag]
CSeq: [cseq] ACK
[last_Call-ID:]
- Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
+ X-Testsuite-Track-Phone-B: 6
Max-Forwards: 70
Content-Length: 0
]]>
@@ -193,13 +197,14 @@
<![CDATA[
BYE sip:[field1]@1[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
- From: <sip:127.0.0.3>
+ From: <sip:127.0.0.3>;tag=[call_number]
To: [$remote_tag]
CSeq: [cseq] BYE
[last_Call-ID:]
- Contact: <sip:[field1]@[local_ip]:[local_port]>
+ Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.3.1734
Accept-Language: en
+ X-Testsuite-Track-Phone-B: 7
Max-Forwards: 70
Content-Length: 0
]]>
diff --git a/tests/fax/pjsip/directmedia_reinvite_t38/test-config.yaml b/tests/fax/pjsip/directmedia_reinvite_t38/test-config.yaml
index 2a21939..2de6b46 100644
--- a/tests/fax/pjsip/directmedia_reinvite_t38/test-config.yaml
+++ b/tests/fax/pjsip/directmedia_reinvite_t38/test-config.yaml
@@ -23,10 +23,15 @@
properties:
minversion: '12.0.0'
dependencies:
- - sipp :
- version : 'v3.0'
- - asterisk : 'chan_pjsip'
- - asterisk : 'res_pjsip_t38'
+ - sipp:
+ version: 'v3.0'
+ - python: 'twisted'
+ - python: 'starpy'
+ - asterisk: 'app_dial'
+ - asterisk: 'app_userevent'
+ - asterisk: 'chan_pjsip'
+ - asterisk: 'res_pjsip'
+ - asterisk: 'res_pjsip_t38'
tags:
- pjsip
- fax
--
To view, visit https://gerrit.asterisk.org/2291
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: merged
Gerrit-Change-Id: Id9e1fac0c8d4d77393f9eca953addabf8c38fcc8
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
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