RTP Payload for DTMF Telephone Events
IVR navigation, call routing, and fax detection depend on RFC 4733 DTMF — not literal G.711 tone audio. Misconfigured `telephone-event` in SDP breaks keypad input on SIP calls.
What It Defines
Carries dual-tone multi-frequency (DTMF) keypad events in RTP using the `telephone-event` dynamic payload type. Event packets include event code (0–15 for 0-9, *, #, A-D), volume, duration, and end-bit — replacing in-band audio tones on VoIP trunks.
Canonical (Normative)
Convenient (Practical)
The canonical publication point for finalized RFCs. If a protocol is standardized as an RFC, the RFC Editor text is the normative final reference. Published by the IETF, IRTF, IAB, and independent stream.
Related Specs
RTP carries all real-time media: VoIP, WebRTC, live video, and video conferencing. Understanding SSRC demultiplexing, jitter buffers, and RTCP feedback loops is essential for any media application.
Codec mismatch debugging starts here — SDP says `PCMU/8000` which is PT 0 per RFC 3551. Static vs dynamic payload types explain most SIP/WebRTC negotiation confusion.
SDP is the handshake language between any two endpoints setting up a call. WebRTC, SIP, and RTSP all use it. You can't debug codec mismatches, ICE failures, or video call setup without reading SDP.
SIP is the PSTN of the internet. Every VoIP platform (Twilio, Vonage, AWS Chime, Asterisk, FreeSWITCH) speaks SIP. Building any communications product requires understanding INVITE flows, registration, and codec negotiation via SDP.
G.711 is the common denominator of telephony. Every PSTN gateway and legacy SIP trunk speaks G.711. When codec negotiation fails and you're left with 64 kbps PCM, this is why.