[asterisk-bugs] [Asterisk 0018314]: Playback() on a meetme/confbridge stutters/drops randomly

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18314 
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Reported By:                mfortini
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18314
Category:                   Applications/app_playback
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           Older 1.6.2 - please test a newer version 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-11-16 03:39 CST
Last Modified:              2010-12-07 04:50 CST
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Summary:                    Playback() on a meetme/confbridge stutters/drops
randomly
Description: 
If I have an extension which does Playback() of a file into a MeetMe or
ConfBridge conference, the playback sound is stuttering or suddenly
stopping randomly.

Capturing the RTP stream, there are no errors like lost or out of order
packets, it's just silence packets.
If I talk on the SIP/somephone which I call on the other extension, the
sound is played back and I can see it on the RTP stream.

I tried it with A* on more than one machine, of varying processing power.

Here's a sample dialplan which triggers the problem:

[default]

exten => 100,1,NoOp()
same  =>     n,Page(SIP/101&SIP/102)
same  =>     n,Hangup()

exten => 101,1,NoOp()
same  =>     n,Dial(SIP/somephone)
same  =>     n,Hangup()

exten => 102,1,NoOp()
same  =>     n,Answer(100)
same  =>     n,Playback(Somelongfile)


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Relationships       ID      Summary
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has duplicate       0018321 Playback() on a meetme/confbridge stutt...
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 (0129384) mfortini (reporter) - 2010-12-07 04:50
 https://issues.asterisk.org/view.php?id=18314#c129384 
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This is the dialplan I used, see the attached logs.

I placed more than one call, and the playback audio stopped at different
times, as I said in the report. If you capture the RTP stream, you can see
it is full of silence packets, not interrupted at all. The calls were all
ended manually afterwards.

exten => 100,1,NoOp()
same  =>           n,Originate(Local/101 at default/n,exten,default,103,1)
same  =>           n,Originate(Local/102 at default/n,exten,default,103,1)
same  =>           n,Wait(2)
same  =>           n,MeetMe(1000,aAtq)
same  =>           n,Hangup()

exten => 101,1,NoOp()
same => n,Dial(SIP/spk02/n)
same => n,Hangup()

exten => 102,1,NoOp()
same => n,Answer(100)
same => n,Playback(longfile8000)

exten => 103,1,NoOp()
same => n,MeetMe(1000,q) 

Issue History 
Date Modified    Username       Field                    Change               
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2010-12-07 04:50 mfortini       Note Added: 0129384                          
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