[asterisk-dev] Jitter Buffer

Pavel Jezek pavel.jezek at i.cz
Mon Oct 23 03:01:32 MST 2006


I tried SIP-IAX call, but still not any info about funtional SIP 
jitterbuffer  :'(
no console SIP jitterbuffer message, nothing in /tmp, nothing in 
asterisk/messages
also it would be nice, if we have consistent jb config (jbenable, 
jbforce etc...) for IAX and other channels
and also some generic way how to debug/monitor jittterbuffer: like "core 
jb debug", instead of iax2 jb debug (or instead of sip jb debug etc.)
PJ

SVN-branch-1.4-r45928M (with jitterbuffer/plc patch: 
http://bugs.digium.com/view.php?id=8189)

my sip.conf

jbenable=yes
jbforce=yes
jbmaxsize=2000
jbimpl=fixed (adaptive tried too)
jblog=yes

iax.conf
jitterbuffer=yes
forcejitterbuffer=yes
maxjitterbuffer=3000
;maxjitterinterps=5
resyncthreshold=-1




John Lange wrote:
> As of Revision 45678 there is a small patch that makes it work with
> jbforce=yes.
>
> However, as far as I can tell it never works unless you force it.
>
> So for example:
>
> PSTN (T1 via Zap) <-> Asterisk 1.4b3 <-> Voip Device
>
> The Asterisk <-> Voip Device leg never gets jitter buffer unless
> forced. 
>
> And yes I'm talking about the received audio at the Asterisk side. It is
> not dejittered.
>
> John
>
> On Fri, 2006-10-20 at 16:11 +0200, Pavel Jezek wrote:
>   
>> I have exactly same problem, no jb log in tmp, no message on console (I 
>> have verbose set to 3),
>> and because sound is very choppy using SIP (much worse than with iax), I 
>> have suspicion, that jitterbuffer is not working/not activated at all.
>> can somebody look to this?
>> PJ
>>
>>
>> jbenable=yes
>> jbforce=yes
>> jbmaxsize=2000
>> jbimpl=fixed (adaptive tried too)
>> jblog=yes
>>
>> Asterisk SVN-branch-1.4-r45678
>>
>>
>>
>>     
>>>  I'm trying to test the new jitter buffer in 1.4beta2 but have not been
>>>       
>> able tell if its working.
>>
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