[asterisk-users] Improvement of PJSIP dtmf_mode description

[Joshua Colp]

On Thu, Aug 3, 2017, at 08:51 AM, Olivier wrote:
> Hello,
> 
> While debugging a SIP trunk with an Avaya IPO, I noticed that wiki's
> PJSIP
> dtmf_mode at [1] includes:
> 
> "This setting allows to choose the DTMF mode for endpoint communication.
> 
>     rfc4733 - DTMF is sent out of band of the main audio stream. This
> supercedes the older RFC-2833 used within the older chan_sip.
>     inband - DTMF is sent as part of audio stream.
>     info - DTMF is sent as SIP INFO packets.
>     auto - DTMF is sent as RFC 4733 if the other side supports it or as
> INBAND if not.
>     auto_info - DTMF is sent as RFC 4733 if the other side supports it or
> as SIP INFO if not."
> 
> 
> The above description doesn't mention anything about incoming DTMF
> treatment.
> May I suggest that:
> - either dtmf_mode has no influence itself on incoming DTMF treatment and
> it could be explicitely mentioned,
> - either dtmf_mode has an influence itself on incoming DTMF treatment and
> this could be described.
> 
> What do you think of this ?

Ultimately we can't explicitly tell a remote side what to use, but the
setting does influence the received somewhat. In the case of rfc4733 we
have to negotiate it in the SDP, and for inband we need to set up a DSP
to listen to the audio stream and detect the DTMF digits. The
documentation could be improved to touch on this some.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org