[asterisk-commits] wdoekes: testsuite/asterisk/trunk r5715 - in /asterisk/trunk/tests/channels/S...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sun Oct 12 03:25:03 CDT 2014
Author: wdoekes
Date: Sun Oct 12 03:24:59 2014
New Revision: 5715
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=5715
Log:
chan_sip: Test unscheduling reINVITE after call hangup.
Belongs with chan_sip commit r425296 (1.8). Tests that no reINVITE is
sent after dialog hangup.
Review: https://reviewboard.asterisk.org/r/4055/
Added:
asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/
asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/
asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/
asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/
asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml (with props)
asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml (with props)
asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/tests.yaml
Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf?view=auto&rev=5715
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf Sun Oct 12 03:24:59 2014
@@ -1,0 +1,2 @@
+[default]
+exten => bob,1,Dial(SIP/bob)
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Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf?view=auto&rev=5715
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf Sun Oct 12 03:24:59 2014
@@ -1,0 +1,21 @@
+[general]
+udpbindaddr=127.0.0.1:5060
+; debugging is nice
+sipdebug=yes
+; we require/expect directmedia reinvites for our test
+directmedia=yes
+; wait 3.2 seconds to timeout retransmitting 200 to re-invite
+; instead of 32 seconds
+timert1=50
+; allow fax
+t38pt_udptl=yes
+
+[alice]
+host=127.0.0.1
+port=5062
+type=friend
+
+[bob]
+host=127.0.0.1
+port=5063
+type=friend
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Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml?view=auto&rev=5715
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml (added)
+++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml Sun Oct 12 03:24:59 2014
@@ -1,0 +1,98 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<!-- Walter Doekes, 2014 for asterisk bug ASTERISK-22791 -->
+<scenario name="ASTERISK-22791-alice">
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[service]@[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=[pid]SIPpTag00[call_number]
+ To: [service] <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:alice@[local_ip]:[local_port]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 [pid][call_number][cseq] [pid][call_number][cseq] IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0 8
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100" optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <recv response="200" rrs="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK [next_url] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ CSeq: 1 ACK
+ Contact: <sip:alice@[local_ip]:[local_port]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- expect directmedia reinvite -->
+ <recv request="INVITE">
+ </recv>
+
+ <!-- reject it, because we were going to hang up -->
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 491 Request Pending
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[pid]SIPpTag01[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:alice@[local_ip]:[local_port]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="ACK">
+ </recv>
+
+ <!-- done with the call -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE [next_url] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: alice <sip:alice@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+ To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ [last_Call-ID:]
+ CSeq: 2 BYE
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+
+ <!-- at this point we *don't* want to see another reINVITE -->
+ <timewait milliseconds="3000"/>
+
+</scenario>
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Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml?view=auto&rev=5715
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml (added)
+++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml Sun Oct 12 03:24:59 2014
@@ -1,0 +1,110 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<!-- Walter Doekes, 2014 for asterisk bug ASTERISK-22791 -->
+<scenario name="ASTERISK-22791-bob">
+
+ <!-- expect call from alice -->
+ <recv request="INVITE">
+ </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:bob@[local_ip]:[local_port]>
+ 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:bob@[local_ip]:[local_port]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 [pid][call_number][cseq] [pid][call_number][cseq] IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv request="ACK">
+ </recv>
+
+ <!-- expect directmedia reinvite -->
+ <recv request="INVITE">
+ </recv>
+
+ <label id="reinvite"/>
+
+ <!-- fine -->
+ <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:bob@[local_ip]:[local_port]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 [pid][call_number][cseq] [pid][call_number][cseq] IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv request="ACK">
+ </recv>
+
+ <recv request="INVITE" optional="true" next="reinvite">
+ </recv>
+
+ <!-- the call gets hung up -->
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- wait a bit to be able to retransmit our 200 -->
+ <timewait milliseconds="3000"/>
+
+</scenario>
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Added: asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml?view=auto&rev=5715
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml Sun Oct 12 03:24:59 2014
@@ -1,0 +1,36 @@
+testinfo:
+ summary: 'Test that we get no reINVITE after 491 after BYE'
+ description: |
+ 'This tests a scenario where asterisk initiates a reINVITE --
+ which gets 491d -- and the call is hung up in the mean time.
+ If the bug is not fixed, we get another reINVITE (with
+ reversed From and To headers). See bug: ASTERISK-22791'
+
+properties:
+ minversion: '1.8.32.0'
+ dependencies:
+ - python: 'starpy'
+ - sipp:
+ version: 'v3.1'
+ - asterisk : 'chan_sip'
+ tags:
+ - SIP
+
+test-modules:
+ test-object:
+ config-section: sipp-config
+ typename: 'sipp.SIPpTestCase'
+
+sipp-config:
+ reactor-timeout: 20
+ fail-on-any: True
+ test-iterations:
+ -
+ scenarios:
+ - {'key-args': {'scenario': 'bob.xml',
+ '-p': '5063',
+ '-default_behaviors': '-bye'}}
+ - {'key-args': {'scenario': 'alice.xml',
+ '-p': '5062',
+ '-s': 'bob',
+ '-default_behaviors': '-bye,abortunexp'}}
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Modified: asterisk/trunk/tests/channels/SIP/tests.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/tests.yaml?view=diff&rev=5715&r1=5714&r2=5715
==============================================================================
--- asterisk/trunk/tests/channels/SIP/tests.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/tests.yaml Sun Oct 12 03:24:59 2014
@@ -71,3 +71,4 @@
- dir: 'ami'
- test: 'invite_retransmit'
- test: 'no_ack_dialog_cleanup'
+ - test: 'no_reinvite_after_491'
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