[svn-commits] mjordan: testsuite/asterisk/trunk r5182 - in /asterisk/trunk/tests/channels/p...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jun 26 12:21:55 CDT 2014
Author: mjordan
Date: Thu Jun 26 12:21:51 2014
New Revision: 5182
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=5182
Log:
channels/pjsip/video_calls: Add a test for fast picture update
This test verifies that when two PJSIP endpoints are in a call with video
enabled, fast picture updates can be received when one endpoint requests
an update via an INFO request.
Review: https://reviewboard.asterisk.org/r/3419/
ASTERISK-23562 #close
Reported by: Matt Jordan
patches:
pjsip_vid_update uploaded by bford (License 6580)
Added:
asterisk/trunk/tests/channels/pjsip/video_calls/
asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/
asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/
asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/
asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/
asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/receive-call.xml (with props)
asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/start-call.xml (with props)
asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/test-config.yaml (with props)
asterisk/trunk/tests/channels/pjsip/video_calls/tests.yaml (with props)
Modified:
asterisk/trunk/tests/channels/pjsip/tests.yaml
Modified: asterisk/trunk/tests/channels/pjsip/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/tests.yaml?view=diff&rev=5182&r1=5181&r2=5182
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/tests.yaml (original)
+++ asterisk/trunk/tests/channels/pjsip/tests.yaml Thu Jun 26 12:21:51 2014
@@ -22,4 +22,4 @@
- dir: 'transfers'
- test: 'info_dtmf'
- test: 'call_pickup'
-
+ - dir: 'video_calls'
Added: asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/extensions.conf?view=auto&rev=5182
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/extensions.conf Thu Jun 26 12:21:51 2014
@@ -1,0 +1,4 @@
+[default]
+
+exten => bob,1,NoOp()
+ same => n,Dial(PJSIP/bob)
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Added: asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/pjsip.conf?view=auto&rev=5182
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/configs/ast1/pjsip.conf Thu Jun 26 12:21:51 2014
@@ -1,0 +1,32 @@
+[local]
+type=transport
+protocol=udp
+bind=0.0.0.0
+
+[endpoint_t](!)
+type=endpoint
+context=default
+transport=local
+direct_media=no
+disallow=all
+allow=ulaw,H261,H263
+
+[aor_t](!)
+type=aor
+max_contacts=1
+
+;; alice
+
+[alice](endpoint_t)
+aors=alice
+
+[alice](aor_t)
+contact=sip:alice at localhost:5061
+
+;; bob
+
+[bob](endpoint_t)
+aors=bob
+
+[bob](aor_t)
+contact=sip:bob at localhost:5062
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Added: asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/receive-call.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/receive-call.xml?view=auto&rev=5182
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/receive-call.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/receive-call.xml Thu Jun 26 12:21:51 2014
@@ -1,0 +1,110 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Receive INVITE with video">
+ <Global variables="remote_tag" />
+ <recv request="INVITE" crlf="true">
+ <!-- Save the from tag. We'll need it when we send our BYE -->
+ <action>
+ <ereg regexp=".*(;tag=.*)"
+ search_in="hdr"
+ header="From:"
+ check_it="true"
+ assign_to="1,remote_tag"/>
+ </action>
+ </recv>
+
+ <Reference variables="1" />
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <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:[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP4 [local_ip]
+ s=-
+ c=IN IP4 [local_ip]
+ t=0 0
+ m=audio 2226 RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ m=video 6002 RTP/AVP 31 32 34
+ a=rtpmap:31 H261/90000
+ a=rtpmap:32 MPV/90000
+ a=rtpmap:34 H263/90000
+ ]]>
+ </send>
+
+ <recv request="ACK" rtd="true" crlf="true">
+ </recv>
+
+ <pause milliseconds="1000"/>
+
+ <send retrans="500">
+ <![CDATA[
+ INFO sip:alice@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:alice@[remote_ip]:[remote_port]>[$remote_tag]
+ [last_Call-ID:]
+ CSeq: [cseq] INFO
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Content-Type: application/media_control+xml
+ Content-Length: [len]
+
+ test
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+
+ <pause milliseconds="1000"/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:alice@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:bob@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:alice@[remote_ip]:[remote_port]>[$remote_tag]
+ [last_Call-ID:]
+ CSeq: [cseq] BYE
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+
+</scenario>
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Added: asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/start-call.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/start-call.xml?view=auto&rev=5182
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/start-call.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/sipp/start-call.xml Thu Jun 26 12:21:51 2014
@@ -1,0 +1,103 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Send INVITE with video to another endpoint">
+ <send retrans="500">
+ <![CDATA[
+ INVITE sip:bob@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+ To: bob <sip:bob@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: [cseq] INVITE
+ Contact: sip:alice@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 1324901698 1324901698 IN IP4 [local_ip]
+ s=-
+ c=IN IP4 [local_ip]
+ t=0 0
+ m=audio 2226 RTP/AVP 0 101
+ a=sendrecv
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ m=video 6002 RTP/AVP 31 32 34
+ a=rtpmap:31 H261/90000
+ a=rtpmap:32 MPV/90000
+ a=rtpmap:34 H263/90000
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="200">
+ </recv>
+
+ <send>
+ <![CDATA[
+ ACK sip:bob@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:alice@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:bob@[remote_ip]:[remote_port]>[peer_tag_param]
+ CSeq: [cseq] ACK
+ Call-ID: [call_id]
+ Contact: <sip:alice@[local_ip]:[local_port]>
+ Allow: INVITE, ACK, MESSAGE, BYE
+ Max-Forwards: 70
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="INFO">
+ <action>
+ <!-- Ensures that the packet is for a fast picture update -->
+ <ereg regexp="application/media_control\+xml"
+ search_in="hdr"
+ header="Content-Type:"
+ check_it="true"
+ assign_to="1"/>
+ </action>
+ </recv>
+
+ <Reference variables="1" />
+
+ <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:alice@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, MESSAGE, BYE
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+ <recv request="BYE" crlf="true" />
+
+ <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:alice@[local_ip]:[local_port];transport=[transport]>
+ Allow: INVITE, ACK, MESSAGE, BYE
+ Content-Type: application/sdp
+ Content-Length: 0
+ ]]>
+ </send>
+
+</scenario>
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Added: asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/test-config.yaml?view=auto&rev=5182
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/video_calls/fast_picture_update/test-config.yaml Thu Jun 26 12:21:51 2014
@@ -1,0 +1,42 @@
+testinfo:
+ summary: 'Tests fast picture updates with video calls using PJSIP.'
+ description:
+ Checks to see if fast picture updates are received during a video call via PJSIP.
+
+test-modules:
+ test-object:
+ config-section: test-object-config
+ typename: 'sipp.SIPpTestCase'
+ modules:
+ -
+ config-section: 'ami-config'
+ typename: 'ami.AMIEventModule'
+
+test-object-config:
+ test-iterations:
+ -
+ scenarios:
+ # Initiates the call
+ - { 'key-args': {'scenario': 'start-call.xml', '-i': '127.0.0.1', '-p': '5061'} }
+ # Receives the call
+ - { 'key-args': {'scenario': 'receive-call.xml', '-i': '127.0.0.1', '-p': '5062'} }
+
+ami-config:
+ -
+ type: 'headermatch'
+ conditions:
+ match:
+ Event: 'TestEvent'
+ State: 'AST_CONTROL_VIDUPDATE'
+ requirements:
+ match:
+ Result: 'Success'
+ count: 1
+
+properties:
+ minversion: '12.4.0'
+ dependencies:
+ - app : 'sipp'
+ - asterisk : 'res_pjsip'
+ tags:
+ - pjsip
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==============================================================================
--- asterisk/trunk/tests/channels/pjsip/video_calls/tests.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/video_calls/tests.yaml Thu Jun 26 12:21:51 2014
@@ -1,0 +1,3 @@
+# Enter tests here in the order they should be considered for execution:
+tests:
+ - test: 'fast_picture_update'
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