Improvement Of PJSIP Dtmf_mode Description

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Asterisk Users 1 Comment

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 ?

Best regards

[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_dtmf_mode

One thought on - Improvement Of PJSIP Dtmf_mode Description

  • 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.