[asterisk-dev] Jitter Buffer

[Pavel Jezek]

maybe good reason to report this issue to bugtracker and publish your 
patch for public... ;-)
PJ


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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