[Asterisk-Users] jitterbuffer and DTMF conflict?

[Martin Joseph]

On Feb 27, 2006, at 6:09 AM, Dr. Michael J. Chudobiak wrote:

> I find that DTMF does not work reliably if jitterbuffer=on for certain 
> IAX providers. For instance, DTMF tones are missed entirely about half 
> the time when I dial into an exgn.net account. However, it always 
> works fine for an unlimitel.ca account.
>
> Someone else has seen this too: http://bugs.digium.com/view.php?id=6011
>
> Can anyone suggest a workaround (other than jitterbuffer=off)?
>
Actually I don't think Asterisk should jitter buffer in the above case?

I am a newb, so be warned.

My research seems to indicate that jitter buffering should only be used 
at the end points, as that is where the audio needs to be reassembled.  
Since in this case asterisk is the man in the middle and not one of the 
endpoints (I think?) it doesn't need to jitter buffer at all for calls 
being placed through an outside IAX carrier? If what I have written is 
true, then jitter buffering is only adding extra latency.

If you are using Zap channels the above is probably wrong though?

I noticed also, from one of my handsets attached to an ATA (AG168v) 
connecting through IAX2, DTMF was sensitive to volume adjustment even 
though it is out of band (rfc2833).

Another thing that might help in the case you describe is to use a more 
band width efficient codec like G729 or GSM versus uLaw or alaw.

Sorry if this is all old hat to you and I am restating the obvious.

Marty