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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4055/
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(Updated Oct. 9, 2014, 9:16 p.m.)
Review request for Asterisk Developers.
Changes
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Previously we would trigger a re-INVITE propagating through Asterisk by
using m=image.
But, Paolo Compagnini was kind enough to report that that doesn't work
with standard sipp with PCAPPLAY enabled.
("media_port keyword with no audio or video on the current line (m=image)")
So, instead we now rely on directmedia to trigger the necessary re-INVITEs
that we're going to delay with a 491.
Bugs: ASTERISK-22791
https://issues.asterisk.org/jira/browse/ASTERISK-22791
Repository: testsuite
Description
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ASTERISK-22791 details how asterisk resends a reINVITE even though the
call has already been hung up by a BYE.
This tests that problem.
Also note how the From/To are also reversed, since this is a reINVITE
*to* alice where alice is in the From.
Diffs (updated)
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/asterisk/trunk/tests/channels/SIP/tests.yaml 5684
/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/test-config.yaml
PRE-CREATION
/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/bob.xml
PRE-CREATION
/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/sipp/alice.xml
PRE-CREATION
/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/sip.conf
PRE-CREATION
/asterisk/trunk/tests/channels/SIP/no_reinvite_after_491/configs/ast1/extensions.conf
PRE-CREATION
Diff: https://reviewboard.asterisk.org/r/4055/diff/
Testing
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Before it is fixed:
<?xml version="1.0" encoding="utf-8"?>
<testsuite errors="0" failures="1" name="AsteriskTestSuite" tests="1"
time="2.84">
<testcase name="tests/channels/SIP/no_reinvite_after_491" time="2.84">
<failure>Running ['./lib/python/asterisk/test_runner.py',
'tests/channels/SIP/no_reinvite_after_491'] ...
[Oct 08 17:54:30] WARNING[4582]: sipp:437 processEnded: Resolving remote host
'127.0.0.1'... Done.
[Oct 08 17:54:30] WARNING[4582]: sipp:437 processEnded: 2014-10-08
17:54:30.202158 1412783670.202158: Aborting call on unexpected message for
Call-Id '[email protected]': while pausing (index 10), received 'INVITE
sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK717504e3;rport
Max-Forwards: 70
From: alice <sip:[email protected]:5062>;tag=4636SIPpTag001
To: bob <sip:[email protected]:5060>;tag=as7d7023cd
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r424181
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 296
v=0
o=root 30542954 30542956 IN IP4 127.0.0.1
s=Asterisk PBX SVN-branch-1.8-r424181
c=IN IP4 127.0.0.1
t=0 0
m=image 4725 udptl t38
c=IN IP4 127.0.0.1
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:389
a=T38FaxUdpEC:t38UDPRedundancy
'.
[Oct 08 17:54:30] WARNING[4582]: sipp:539 __scenario_callback: SIPp Scenario
alice.xml Failed [1]
[Oct 08 17:54:30] WARNING[4582]: sipp:548 __evaluate_scenario_results: SIPp
Scenario alice.xml Failed
[Oct 08 17:54:30] WARNING[4582]: sipp:402 kill: Killing SIPp Scenario bob.xml
</failure>
</testcase>
</testsuite>
After a possible fix:
<?xml version="1.0" encoding="utf-8"?>
<testsuite errors="0" failures="0" name="AsteriskTestSuite" tests="1"
time="9.95">
<testcase name="tests/channels/SIP/no_reinvite_after_491" time="9.95"/>
</testsuite>
Thanks,
wdoekes
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