How DTMF are sent in VoIP

I was checking today at how SIP phones negotiate DTMF tones transmission. It looks like most VoIP applications including Asterix transfer DTMF using 101 payload type. Guys at MS say here that Cisco also has a default value of 101 as payload type for DTMF out of band transmission
However, this payload type is not always set explicitly for “telephone-event” codec during SDP session negotiation.

For example when Nero SIPS software phone 2.1.3 sends INVITE or OK it does not specify how it will send DTMF but uses 101 payload type.

What surprised me is that 101 is actually a dynamic code so that means applications should
negotiate the codecs using SDP.
However, sometimes DTMP data description is not included in SDP.

I did not have a chance to test this on all SIP phones, however Nero SIPS seems to have this issue.

By the way RFC 2833 which explains payload for telephone events does not explicitly specify the type code.

RFC 2833:
In accordance with current practice, this payload format does not have a static payload type number, but uses a RTP payload type number established dynamically and out-of-band.


I was wondering is there any standard document which defines 101 as default DTMF payload type.

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