[svn-commits] wdoekes: testsuite/asterisk/trunk r5709 - in /asterisk/trunk/tests/channels/S...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Oct 10 02:25:26 CDT 2014
Author: wdoekes
Date: Fri Oct 10 02:25:08 2014
New Revision: 5709
URL: http://svnview.digium.com/svn/testsuite?view=rev&rev=5709
Log:
Test chan_sip dialog leak reported in ASTERISK-20784.
Under the right circumstances, a leg never gets destroyed, resulting in
file leaks and other nastiness.
Fix committed in asterisk 1.8 r425068.
Added:
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/extensions.conf (with props)
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/sip.conf (with props)
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/run-test (with props)
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/alice.xml (with props)
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/bob.xml (with props)
asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/test-config.yaml (with props)
Modified:
asterisk/trunk/tests/channels/SIP/tests.yaml
Added: asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/extensions.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/extensions.conf?view=auto&rev=5709
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/extensions.conf (added)
+++ asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/extensions.conf Fri Oct 10 02:25:08 2014
@@ -1,0 +1,2 @@
+[default]
+exten => bob,1,Dial(SIP/bob)
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Added: asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/sip.conf
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/sip.conf?view=auto&rev=5709
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/sip.conf (added)
+++ asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/configs/ast1/sip.conf Fri Oct 10 02:25:08 2014
@@ -1,0 +1,19 @@
+[general]
+udpbindaddr=127.0.0.1:5060
+; debugging is nice
+sipdebug=yes
+; don't do any re-invites, it messes up our scenario
+directmedia=no
+; wait 3.2 seconds to timeout retransmitting 200 to re-invite
+; instead of 32 seconds
+timert1=50
+
+[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_ack_dialog_cleanup/run-test
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/run-test?view=auto&rev=5709
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/run-test (added)
+++ asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/run-test Fri Oct 10 02:25:08 2014
@@ -1,0 +1,156 @@
+#!/usr/bin/env python
+'''
+Copyright (C) 2014, Digium, Inc.
+Walter Doekes <walter+asterisk at wjd.nu>
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+
+
+Bug ASTERISK-20784:
+
+ A INVITE> asterisk INVITE> B
+ A <200 OK asterisk <200 OK B
+ A -ACK--> asterisk -ACK--> B
+ asterisk <INVITE B
+ asterisk 200 OK> B
+ (ack lost)
+ A -BYE--> asterisk
+ A <200 OK asterisk
+
+At this point the B leg retransmit the 200 to the INVITE and finally
+leaks a dialog object. Or at least, it did until this bug was fixed.
+
+We cannot just use the yaml-based sipp.SIPpTestCase config, because
+we need to extract a bit of extra info from Asterisk after the
+scenario's complete.
+'''
+
+import logging
+import os
+import sys
+import time
+
+sys.path.append("lib/python")
+sys.path.append("utils")
+
+from twisted.internet import reactor
+from asterisk.sipp import SIPpTest
+
+WORKING_DIR = os.path.abspath(os.path.dirname(__file__))
+TEST_DIR = os.path.dirname(os.path.realpath(__file__))
+
+LOGGER = logging.getLogger(__name__)
+
+
+class MySIPpTest(SIPpTest):
+ def __init__(self):
+ # A list of jobs to complete before stopping completely. This
+ # list is extended during the stoppage.
+ self.stop_jobs = [self.get_sleep,
+ self.get_channels,
+ self.get_sip_objects]
+
+ SIPP_SCENARIOS = [
+ {'scenario': 'bob.xml',
+ '-p': '5063',
+ '-default_behaviors': '-bye'},
+ {'scenario': 'alice.xml',
+ '-p': '5062',
+ '-s': 'bob',
+ '-default_behaviors': '-bye'},
+ ]
+
+ super(MySIPpTest, self).__init__(WORKING_DIR, TEST_DIR, SIPP_SCENARIOS)
+
+ def clean_cli_output(self, output):
+ if output.endswith('Asterisk ending (0).\n'):
+ output = output[0:-21]
+ return output.strip()
+
+ def should_be_done_by_now(self):
+ # We use a timert1 of 50 which should stop retransmissions
+ # after 3.2 seconds.
+ # t0 = call is initiated
+ # t1 = call is complete (except for Bob-leg)
+ # t5 = asterisk has timed out sending 200s, start sending BYE
+ # t9 = asterisk has timed out sending BYE, destroy object
+ # (in practice, it turns out it takes about 13+ seconds)
+ # t20 = give everyone some extra time
+ return (time.time() - self.t0 > 20)
+
+ def get_channels(self):
+ def ret(obj):
+ output = self.clean_cli_output(obj.output)
+ if output.startswith('0 active channels'):
+ # No need to schedule ourselves anymore.
+ pass
+ elif self.should_be_done_by_now():
+ LOGGER.warn('Expected 0 active calls: %r' % (output,))
+ self.passed = False
+ else:
+ # Try again in a while.
+ self.stop_jobs.extend([self.get_sleep, self.get_channels])
+ self.next_job()
+ self.ast[0].cli_exec('core show channels count').addCallback(ret)
+
+ def get_sip_objects(self):
+ def ret(obj):
+ output = self.clean_cli_output(obj.output)
+ dialog_objs = output.split('-= Dialog objects:', 1)[-1].strip()
+ if not dialog_objs:
+ # No need to schedule ourselves anymore.
+ pass
+ elif self.should_be_done_by_now():
+ LOGGER.warn('Expected 0 sip dialog objects:\n %s' %
+ (dialog_objs.replace('\n', '\n '),))
+ self.passed = False
+ else:
+ # Try again in a while.
+ self.stop_jobs.extend([self.get_sleep, self.get_sip_objects])
+ self.next_job()
+ self.ast[0].cli_exec('sip show objects').addCallback(ret)
+
+ def get_sleep(self):
+ reactor.callLater(1, self.next_job)
+
+ def next_job(self):
+ if not self.stop_jobs:
+ # Really done.
+ super(MySIPpTest, self).stop_reactor()
+ else:
+ # Schedule the next job.
+ first_job = self.stop_jobs.pop(0)
+ first_job()
+
+ def run(self):
+ """
+ This is called by the TestCase when we start. Start counting
+ the clock.
+ """
+ super(MySIPpTest, self).run()
+ self.t0 = time.time()
+
+ def stop_reactor(self):
+ """
+ This is called by the TestCase when we're done. Defer that
+ until after we've pulled some info out of asterisk.
+ """
+ self.next_job()
+
+
+def main():
+ test = MySIPpTest()
+ reactor.run()
+
+ # If it failed, bail.
+ if not test.passed:
+ return 1
+
+ return 0
+
+
+if __name__ == "__main__":
+ sys.exit(main())
+
+# vim:sw=4:ts=4:expandtab:textwidth=79
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Added: asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/alice.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/alice.xml?view=auto&rev=5709
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/alice.xml (added)
+++ asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/alice.xml Fri Oct 10 02:25:08 2014
@@ -1,0 +1,75 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="ASTERISK-20784-alice">
+
+ <!-- invite with ulaw -->
+ <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
+ 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>
+
+ <!-- make sure bob is first with his reinvite -->
+ <pause milliseconds="100"/>
+
+ <!-- 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: 3 BYE
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+
+</scenario>
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Added: asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/bob.xml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/bob.xml?view=auto&rev=5709
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/bob.xml (added)
+++ asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/sipp/bob.xml Fri Oct 10 02:25:08 2014
@@ -1,0 +1,94 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="ASTERISK-20784-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">
+ <action>
+ <ereg regexp="<(.*)>" search_in="hdr" header="Contact:" assign_to="_,contact"/>
+ <ereg regexp=".*" search_in="hdr" header="From:" assign_to="from"/>
+ <ereg regexp=".*" search_in="hdr" header="To:" assign_to="to"/>
+ </action>
+ </recv>
+ <Reference variables="_"/>
+
+ <!-- re-invite with alaw -->
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE [$contact] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: [$to]
+ To: [$from]
+ Call-ID: [call_id]
+ CSeq: 2 INVITE
+ 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 8
+ a=rtpmap:8 PCMA/8000
+
+ ]]>
+ </send>
+
+ <!-- expect hung up call -->
+ <recv request="BYE" optional="true">
+ </recv>
+
+ <!-- expected response to our reinvite -->
+ <recv response="100" optional="true">
+ </recv>
+
+ <!-- expected response to our reinvite -->
+ <recv response="200">
+ </recv>
+
+</scenario>
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Added: asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/test-config.yaml
URL: http://svnview.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/test-config.yaml?view=auto&rev=5709
==============================================================================
--- asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/test-config.yaml (added)
+++ asterisk/trunk/tests/channels/SIP/no_ack_dialog_cleanup/test-config.yaml Fri Oct 10 02:25:08 2014
@@ -1,0 +1,17 @@
+testinfo:
+ summary: 'Test channel leak as described in ASTERISK-20784'
+ description: |
+ 'This tests a scenario where Alice calls Bob through Asterisk.
+ Alice hangs up immediately. But Bob attempts a re-INVITE at
+ the same time. At this point Bob stops to respond.
+ If the bug is not fixed, this will leak a sip dialog object.'
+
+properties:
+ minversion: '1.8.0.0'
+ dependencies:
+ - python: 'starpy'
+ - sipp:
+ version: 'v3.1'
+ - asterisk : 'chan_sip'
+ tags:
+ - SIP
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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=5709&r1=5708&r2=5709
==============================================================================
--- asterisk/trunk/tests/channels/SIP/tests.yaml (original)
+++ asterisk/trunk/tests/channels/SIP/tests.yaml Fri Oct 10 02:25:08 2014
@@ -70,3 +70,4 @@
- test: 'tel_uri'
- dir: 'ami'
- test: 'invite_retransmit'
+ - test: 'no_ack_dialog_cleanup'
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