DTMF Digits Transported in RFC 2833

DTMF Digits Transported in RFC 2833

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How are DTMF digits transported in RFC 2833?



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A. B. C. D. E.


DTMF digits and named telephone events are carried as part of the audio stream, and MUST use the same sequence number and time-stamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway.

The default clock frequency is 8,000 Hz, but the clock frequency can be redefined when assigning the dynamic payload type.

RFC 2833 (Remote Digital Audio for Telephony) is a protocol used to transmit dual-tone multi-frequency (DTMF) signaling, which is the audible tones generated by a telephone keypad, in Voice over IP (VoIP) networks. It was developed to overcome the limitations of in-band DTMF, where the DTMF tones are sent along with the audio stream, causing potential issues with audio quality and reliability.

Answer: A. In the RTP stream with the named telephone events payload format.

In RFC 2833, DTMF digits are transported in the RTP (Real-time Transport Protocol) stream using the Named Telephone Events (NTE) payload format. This payload format allows for the transmission of DTMF events as separate RTP packets, instead of in-band signaling, thereby ensuring their accurate and reliable delivery.

The NTE payload format uses a special RTP payload type number (101) to indicate that the payload contains DTMF event data. The payload contains a DTMF event header followed by the DTMF digits. The DTMF event header provides information about the duration of the DTMF tone, the DTMF digit, and whether the tone is starting or ending.

The RTP stream with the NTE payload format is typically used in conjunction with Session Initiation Protocol (SIP) or H.323 signaling protocols. When a caller enters a DTMF digit, the DTMF event is sent in an RTP packet with the NTE payload format to the recipient over the IP network.

In summary, RFC 2833 uses the NTE payload format in the RTP stream to transport DTMF digits in a separate packet to ensure their accurate and reliable delivery in VoIP networks.