Perhaps I have not stated the problem well. When a Media Gateway
receives a call from the PSTN and it is using RFC2833/4733 for DTMF, it
typically suppresses the sending of the DTMF tone as encoded audio and
sends only the RTP event. It takes some time for the Media Gateway to
actually detect the DTMF tone before it can suppress the encoded audio.
During that time, some of the DTMF tone can leak into the encoded audio.
The leakage can be reduced by additional buffering but this in turn adds
latency. I'm trying to determine if there is a generally acceptable
range for this leakage. I have seen values ranging from 5ms which is not
too significant, up to 50ms, which is quite significant and which
creates problems for some VoIP applications.

Thanks,
James

-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, May 02, 2007 9:16 PM
To: [email protected]
Subject: Re: [Sip-implementors] RFC2833 Implementations - Inband Leakage
vs. Latency ?

   From: [EMAIL PROTECTED]

   In case of DTMF digits, a single RFC 4733 packet carries a single
DTMF
   digit in my understanding. Triple redundancy is the basic means to
target
   RTP packet losses, i.e. the receiver must receive at least one out of
   three.

Be careful -- An RFC 4733 packet carries state information about a
DTMF sound, either that the sound is on (and if so, how long it has
been on) or that the sound has stopped.  E.g., it is up to the
recipient to decode this information into DTMF digits, and in some
cases, to distinguish how long the digit was.  Multiple packets are
used to defeat packet loss, but also to allow transmitting the
initiation of the digit before the finish of the digit has been seen,
and to allow the transmission of the real length of the digit.

Dale
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