[asterisk-dev] Jitter Buffer

Pavel Jezek pavel.jezek at i.cz
Mon Oct 23 08:11:05 MST 2006



John Lange wrote:
> On Mon, 2006-10-23 at 12:01 +0200, Pavel Jezek wrote:
>   
>> 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
>>     
>
> There was a patch recently (after 1.4b3 came out) that fixes this.
>
>   

I'm using 1.4 branch rev45928 (checked out today), this mean, it should 
contain this patch...
>> also it would be nice, if we have consistent jb config (jbenable, 
>> jbforce etc...) for IAX and other channels
>>     
>
> It is consistent. Use the same syntax in all files. What you are using
> below for IAX is the old jitterbuffer.
>
>   
I set this options from iax.conf.sample (from configs dir), that is 
probably obsolete

>> 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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>>>>         
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>
>
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