[Asterisk-code-review] testsuite: Cleanup tests/fax/sip/directmedia reinvite t38 (testsuite[master])
Richard Mudgett
asteriskteam at digium.com
Tue Feb 23 18:46:11 CST 2016
Richard Mudgett has uploaded a new change for review.
https://gerrit.asterisk.org/2292
Change subject: testsuite: Cleanup tests/fax/sip/directmedia_reinvite_t38
......................................................................
testsuite: Cleanup tests/fax/sip/directmedia_reinvite_t38
Basically this is taking the latest files from
tests/fax/pjsip/directmedia_reinvite_t38 and adapting them back to this
test.
* 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.
Change-Id: If8daa4e8cba2f7fc8f2b1b3ef33840d2815b5f27
---
M tests/fax/sip/directmedia_reinvite_t38/configs/ast1/extensions.conf
M tests/fax/sip/directmedia_reinvite_t38/sipp/endpoint_A.xml
M tests/fax/sip/directmedia_reinvite_t38/sipp/endpoint_B.xml
M tests/fax/sip/directmedia_reinvite_t38/test-config.yaml
4 files changed, 48 insertions(+), 78 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/92/2292/1
diff --git a/tests/fax/sip/directmedia_reinvite_t38/configs/ast1/extensions.conf b/tests/fax/sip/directmedia_reinvite_t38/configs/ast1/extensions.conf
index 5405e4b..208ec18 100644
--- a/tests/fax/sip/directmedia_reinvite_t38/configs/ast1/extensions.conf
+++ b/tests/fax/sip/directmedia_reinvite_t38/configs/ast1/extensions.conf
@@ -1,13 +1,4 @@
-[general]
-PHONE_TO_DIAL=SIP/endpoint_B
-
[default]
-exten => bypassbridge,1,NoOp()
- same => n,Dial(SIP/endpoint_B,,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(SIP/endpoint_B,,g)
same => n,UserEvent(TestStatus, extension: basicdial)
diff --git a/tests/fax/sip/directmedia_reinvite_t38/sipp/endpoint_A.xml b/tests/fax/sip/directmedia_reinvite_t38/sipp/endpoint_A.xml
index 47db1f8..796e322 100644
--- a/tests/fax/sip/directmedia_reinvite_t38/sipp/endpoint_A.xml
+++ b/tests/fax/sip/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]
@@ -108,27 +110,14 @@
<recv request="ACK"/>
<!-- Reinvite received for T38 - media flows between Enpoint A and Asterisk -->
- <recv request="INVITE">
- <action>
- <ereg regexp="c=IN IP4 127.0.0.1" search_in="body" check_it="true" assign_to="1" />
- <log message="Side A - Contact SDP for T38 reinvite matches expectations: [$1]." />
- <strcmp assign_to="result" variable="1" value="c=IN IP4 127.0.0.1" />
- <test assign_to="emptyinv2" variable="result" compare="not_equal" value="" />
- </action>
- </recv>
-
- <nop condexec="emptyinv2">
- <action>
- <error message="Side A - Contact SDP for T38 reinvite did not match - expected 'c=IN IP4 127.0.0.1' but got [$1]" />
- </action>
- </nop>
+ <recv request="INVITE" />
<send retrans="500">
<![CDATA[
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]>
@@ -136,7 +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]
@@ -166,7 +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]>
@@ -174,20 +163,20 @@
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]
v=0
- o=- 1324901698 1324901700 IN IP4 [local_ip]
+ o=- 1324901698 1324901698 IN IP4 [local_ip]
s=Polycom IP Phone
c=IN IP4 [local_ip]
t=0 0
- m=image 10972 udptl t38
a=sendrecv
- a=T38FaxVersion:0
- a=T38MaxBitRate:9600
- a=T38FaxUdpEC:t38UDPRedundancy
+ m=audio 2226 RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
]]>
</send>
@@ -200,16 +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/sip/directmedia_reinvite_t38/sipp/endpoint_B.xml b/tests/fax/sip/directmedia_reinvite_t38/sipp/endpoint_B.xml
index 4ef7a27..4781ba0 100644
--- a/tests/fax/sip/directmedia_reinvite_t38/sipp/endpoint_B.xml
+++ b/tests/fax/sip/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>
@@ -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]
@@ -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]
@@ -133,16 +135,17 @@
<!-- Reinvite to set up T38 Fax session -->
<send retrans="500">
<![CDATA[
- INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
+ INVITE sip:endpoint_B@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
- From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
- To: [field0] <sip:[field1]@[remote_ip]>
+ 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
@@ -167,34 +170,21 @@
<recv response="100" optional="true" />
- <recv response="200">
- <action>
- <ereg regexp="c=IN IP4 127.0.0.1" search_in="body" check_it="true" assign_to="1" />
- <log message="Side B - Contact SDP for T38 reinvite 200 OK matches expectations: [$1]." />
- <strcmp assign_to="result" variable="1" value="c=IN IP4 127.0.0.1" />
- <test assign_to="empty200" variable="result" compare="not_equal" value="" />
- </action>
- </recv>
-
- <nop condexec="empty200">
- <action>
- <error message="Side B - Contact SDP for T38 reinvite 200 OK did not match - expected 'c=IN IP4 127.0.0.1' but got [$1]" />
- </action>
- </nop>
-
+ <recv response="200" />
<send>
<![CDATA[
ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
- From: [field1] <sip:[field1]@[local_ip]>;tag=[call_number]
- To: <sip:[field0]@[remote_ip];user=[field0]>[peer_tag_param]
+ From: <sip:127.0.0.3>;tag=[call_number]
+ To: [$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
]]>
@@ -207,18 +197,20 @@
<![CDATA[
BYE sip:[field1]@1[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch]
- From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
- To: [field0] <sip:[field1]@[remote_ip]>[$remote_tag]
+ From: <sip:127.0.0.3>;tag=[call_number]
+ To: [$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
]]>
</send>
+ <recv response="200" />
</scenario>
diff --git a/tests/fax/sip/directmedia_reinvite_t38/test-config.yaml b/tests/fax/sip/directmedia_reinvite_t38/test-config.yaml
index cde33b5..09cb8cd 100644
--- a/tests/fax/sip/directmedia_reinvite_t38/test-config.yaml
+++ b/tests/fax/sip/directmedia_reinvite_t38/test-config.yaml
@@ -23,10 +23,10 @@
properties:
minversion: '1.8.0.0'
dependencies:
- - sipp :
- version : 'v3.0'
- - python : 'twisted'
- - python : 'starpy'
+ - sipp:
+ version: 'v3.0'
+ - python: 'twisted'
+ - python: 'starpy'
- asterisk: 'app_dial'
- asterisk: 'app_userevent'
- asterisk: 'chan_sip'
--
To view, visit https://gerrit.asterisk.org/2292
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: If8daa4e8cba2f7fc8f2b1b3ef33840d2815b5f27
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
More information about the asterisk-code-review
mailing list