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List:       asterisk-dev
Subject:    Re: [asterisk-dev] SIP trunk DTMF audio bleeding -- any way to
From:       Benny Amorsen <benny+usenet () amorsen ! dk>
Date:       2011-07-27 8:34:57
Message-ID: m339hsqfym.fsf () ursa ! amorsen ! dk
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Kirill Katsnelson <kkm@adaptiveai.com> writes:

> We have a problem with DTMF tones bleeding as short tones when
> received from the trunk provider. First, there is a batch of RFC
> whatever the number is DTMF payload messages in RTP stream, and then a
> 30-50 ms burst of the matching audio tone. In other words, the
> provider does not mask out the tone completely, and allows a tail to
> come through. These tones give us some trouble.
>
> There are multiple ways to resolve that, but I am trying to start with
> the simplest approach: the configuration. Is there a setting in
> Asterisk that would allow me to strip these audio tails? I. e. mute
> the channel and drop incoming audio (or replace with a silence) for N
> ms after the last DTMF RTP message received?

Not as far as I know.

> And, if not, would such a feature be of general interest?

It certainly sounds useful. However, it would be difficult to apply to
DTMF in SIP INFO, and it would be tricky to do it with fast-switched
or reinvited RTP.

The additional issue you will hit is that Asterisk is not transparent
for DTMF unless it does fast-switching or reinvites. That is, the length
of the DTMF tone will be changed. You can perhaps use that undocumented
feature to your advantage -- try adding tT to the Dial command for a
test call and see if the problem changes. Obviously tT isn't a good
idea in production unless you actually want to provide that feature.


/Benny


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