[asterisk-dev] Jitter Buffer

[John Lange]

Have a look at the other thread on this topic. This new JB
implementation seems thoroughly confusing to me at this point.

>From what I understand, if you are doing SIP <-> IAX and you want the JB
on the IAX leg you have to enable it in sip.conf(?!).

In my case I need the JB on the SIP leg of a SIP<->ZAP call so I have to
enable it in zapata.conf.

Can someone bring some clarity to the issue?

BTW, there is a bug in the JB which causes no audio to flow in one
direction. I'm trying to track it down and will file a bug report if I
can isolate it.

John

On Mon, 2006-10-23 at 17:11 +0200, Pavel Jezek wrote:
> 
> 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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