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

[Martin Joseph]

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