[Asterisk-bugs] [Asterisk 0009631]: improve jitterbuffer (jbtargetextra/minjitterbuffer)

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=9631 
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Reported By:                pj
Assigned To:                
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Project:                    Asterisk
Issue ID:                   9631
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   tweak
Priority:                   normal
Status:                     feedback
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 61800 
Disclaimer on File?:        No 
Request Review:              
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Date Submitted:             04-29-2007 06:32 CDT
Last Modified:              07-19-2007 11:15 CDT
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Summary:                    improve jitterbuffer (jbtargetextra/minjitterbuffer)
Description: 
I discovered, that jittertargetextra option works maybe not good as it
can,
eg. if I have jitter spike (rtt 5ms to 100ms) and jittertargetextra set to
150ms, so that total jb size (155ms) can compensate this (95ms) jitter
change,
but nagative aspect of how jittertargetextra is now implemented is, that
it _immediatelly_ extend jitterbuffer from 155ms to 250ms!
so it have all negative consequenses to voice quality as "primary" jitter
spike.
this can be improved by growing jitterbuffer size _gently_ and not make jb
size jumps.
I think also, that it will be really usefull to add similar concept like
maxjitterbuffer, to be able to specify also _minimum_ jb size, that jb size
shouldn't never drop below lower limit, regardless of actual measured
jitter.
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---------------------------------------------------------------------- 
 stevedavies - 07-19-07 11:15  
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Please don't apply this one without discussing with Steve Kann and me.
A jitter spike *should* result in audio being lost.  That's presuming its
really a spike.  If we confuse ourselves into thinking that the jitter
buffer is there to prevent all audio disruption, we'll end up with huge
jitter buffers and associated long delay.  dejittering is a balancing act -
if jitter on a call is usually say 20msec and we have say 1 1-second
dropout per minute then the jitter buffer should be set to handle the
20msec jitter and the 1 second dropouts should result in audio dropouts. 
Otherwise we have to run a 1000msec jitter buffer all the time which will
be much much worse than the single dropout.

Steve 

Issue History 
Date Modified   Username       Field                    Change               
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07-19-07 11:15  stevedavies    Note Added: 0067599                          
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