[asterisk-dev] Re: [asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad

[Anton]

I have the same issue. Just conditions does not depend to 
jittery link. It just goes one-way after some while. 

http://bugs.digium.com/view.php?id=8273

Looks like noone interested in that... even considering that 
bug is critical.

On 11 November 2006 00:24, Pavel Jezek wrote:
> if anyone has one-way audio issues with iax over jittery
> connection, please look at bug report, what I created
> yesterday and report your experiences,
> I think this is one of the most serious bug, that must be
> identified and resolved before 1.4 will be released,
> thanks
> http://bugs.digium.com/view.php?id=8325
> PJ
>
> Benjamin Jacob wrote:
> > Martin Joseph wrote:
> >> On 2006-10-25 08:14:43 -0700, "Noah Miller"
> >>
> >> <noahisaacmiller at gmail.com> said:
> >>> Hi Matt -
> >>>
> >>>> I have a customer who experiences, once in a while,
> >>>> one-way audio... That is... they can hear the person
> >>>> they called, but the person can not hear them.
> >>>>
> >>>> On the customer's end I have the following config in
> >>>> iax.conf: trunk=no
> >>>> (I have also tried trunk=yes and nothing for trunk=)
> >>>> jitterbuffer=yes
> >>>> forcejitterbuffer=yes
> >>>> dropcount=3
> >>>> minexcessbuffer=80
> >>>> jittershrinkrate=1
> >>>
> >>> If you're using Asterisk 1.2.x, dropcount,
> >>> jittershrinkrate and minexcesbuffer don't do
> >>> anything.  They are ignored by 1.2.x unless you
> >>> specify that you want to use the old 1.0.x
> >>> jitterbuffer.  Instead you might try the parameters
> >>> maxjitterbuffer, resyncthreshold, and
> >>> maxjitterinterps.  For more, you can check out the
> >>> sample iax.conf.
> >>>
> >>> I believe, also, that you are correct in setting
> >>> trunk=no.  I know in the 1.0.x jitterbuffer, trunk
> >>> was not fully supported.  I think this is still the
> >>> case with the 1.2.x jitterbuffer.
> >>
> >> If the audio is dropping out completely, then I
> >> suspect the whole jitter buffer thing is a red herring
> >> (waste of time).
> >>
> >> Perhaps it's a nat issue?  What kind of router if any
> >> is involved?  I am reaching here... Also, please do
> >> tell us which version of asterisk you are running...
> >>
> >> Marty
> >
> > seeing this thread a lil too late, i guess. So, am
> > sorry if I am repeating things.
> > When I was setting up my iax2 configs, I too had one
> > way audio initialy. Tried the softphone on two
> > machines(which incidentaly had asterisk running on them
> > as well), to no avail. When I looked at the tcpdump on
> > my asterisk server, I could see no rtp coming in from
> > the two said machines.
> > So, I shifted the softphone to another machine, this
> > time on a windows machine, n voila! it worked like a
> > charm.
> >
> > So, I hope you did have a look at the tcpdump to check
> > on the rtp flow.
> >
> > cheerz
> > - Ben.
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