[asterisk-commits] kharwell: testsuite/asterisk/trunk r6120 - in /asterisk/trunk/tests/channels/...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Dec 18 09:55:14 CST 2014
Author: kharwell
Date: Thu Dec 18 09:55:10 2014
New Revision: 6120
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=6120
Log:
res_pjsip_sdp_rtp: Incompatible DTMF mode test
Checks to make sure that if two pjsip endpoints with incompatible DTMF modes
are connected the appropriate bridge type is chosen.
ASTERISK-24459
Reported by: Yaniv Simhi
Review: https://reviewboard.asterisk.org/r/4266/
Added:
asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/
asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/
asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/
asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/pjsip.conf (with props)
asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/
asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/invite.xml (with props)
asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/invite_recv.xml (with props)
asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/pjsip/tests.yaml
Added: asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/extensions.conf?view=auto&rev=6120
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/extensions.conf Thu Dec 18 09:55:10 2014
@@ -1,0 +1,3 @@
+[default]
+exten => bob,1,Dial(PJSIP/${EXTEN})
+ same => n,Hangup()
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Added: asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/pjsip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/pjsip.conf?view=auto&rev=6120
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/pjsip.conf (added)
+++ asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/configs/ast1/pjsip.conf Thu Dec 18 09:55:10 2014
@@ -1,0 +1,39 @@
+[local]
+type=transport
+bind=0.0.0.0
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[endpoint_t](!)
+type=endpoint
+context=default
+transport=local
+direct_media=no
+disallow=all
+allow=ulaw
+
+[aor_t](!)
+type=aor
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+;;; alice
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[alice](aor_t)
+contact=sip:alice at localhost:5061
+
+[alice](endpoint_t)
+aors=alice
+dtmf_mode=inband
+
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+;;; bob
+;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
+
+[bob](aor_t)
+contact=sip:bob at localhost:5062
+
+[bob](endpoint_t)
+aors=bob
+dtmf_mode=rfc4733
+
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svn:keywords = Author Date Id Rev URL
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svn:mime-type = text/plain
Added: asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/invite.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/invite.xml?view=auto&rev=6120
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/invite.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/invite.xml Thu Dec 18 09:55:10 2014
@@ -1,0 +1,71 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Invite with variables">
+ <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
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true" />
+
+ <recv response="180" optional="true" />
+
+ <recv response="200" />
+
+ <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]>;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]>
+ Allow: INVITE, ACK, MESSAGE, BYE
+ Max-Forwards: 70
+ 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/dtmf_incompatible/sipp/invite_recv.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/invite_recv.xml?view=auto&rev=6120
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/invite_recv.xml (added)
+++ asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/sipp/invite_recv.xml Thu Dec 18 09:55:10 2014
@@ -1,0 +1,77 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Invite with variables">
+ <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=.*)"
+ header="From:"
+ search_in="hdr"
+ check_it="true"
+ assign_to="remote_tag"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[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=[pid]SIPpTag01[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
+ ]]>
+ </send>
+
+ <recv request="ACK" rtd="true" crlf="true" />
+
+ <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=[pid]SIPpTag01[call_number]
+ To: [$remote_tag]
+ [last_Call-ID:]
+ CSeq: [cseq] BYE
+ Contact: sip:bob@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" />
+</scenario>
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Added: asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/test-config.yaml?view=auto&rev=6120
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/pjsip/dtmf_incompatible/test-config.yaml Thu Dec 18 09:55:10 2014
@@ -1,0 +1,46 @@
+testinfo:
+ summary: 'Test incompatible DTMF modes.'
+ description: |
+ 'Checks to make sure that if two endpoints with incompatible DTMF modes
+ are connected the appropriate bridge type is chosen.'
+
+properties:
+ minversion: '13.2.0'
+ dependencies:
+ - app : 'sipp'
+ - asterisk : 'res_pjsip'
+ tags:
+ - 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:
+ - { 'key-args': { 'scenario':'invite_recv.xml', '-p':'5062' } }
+ - { 'key-args': { 'scenario':'invite.xml', '-p':'5061' } }
+
+ami-config:
+ -
+ type: 'headermatch'
+ conditions:
+ match:
+ Event: 'BridgeLeave'
+ BridgeTechnology: 'simple_bridge'
+ count: 2 # two channels in, two out
+ -
+ type: 'headermatch'
+ conditions:
+ match:
+ Event: 'BridgeLeave'
+ BridgeTechnology: 'native_bridge'
+ count: 0 # should never bridge natively
+
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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=6120&r1=6119&r2=6120
==============================================================================
--- asterisk/trunk/tests/channels/pjsip/tests.yaml (original)
+++ asterisk/trunk/tests/channels/pjsip/tests.yaml Thu Dec 18 09:55:10 2014
@@ -29,3 +29,4 @@
- dir: 'config_wizard'
- dir: 'optimistic_srtp'
- test: 'in_dialog_invite_replaces'
+ - test: 'dtmf_incompatible'
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