Real-time Transport Protocol
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.
What It Defines
RTP provides end-to-end delivery of real-time audio and video over UDP. Defines the 12-byte header (version, padding, extension, CC, marker, payload type, sequence number, timestamp, SSRC, CSRC), payload format negotiation, and the companion RTCP protocol for sender/receiver reports, NACK, and jitter/loss statistics.
Canonical (Normative)
Related References
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
Any media you send over the internet should be encrypted. SRTP is the protocol. In WebRTC it's automatic via DTLS-SRTP; in SIP deployments you must configure it explicitly. Know it when auditing VoIP security.
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.
Opus is the codec for real-time audio on the internet. WebRTC mandates it; Discord, Zoom, and most VoIP apps use it. Understanding bitrate, FEC, and frame sizing directly impacts audio quality and bandwidth.
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.
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.
WebRTC and modern SIP use RFC 7587 Opus RTP framing — not Ogg pages. FEC and DTX SDP attributes directly control call quality under packet loss.
WAV is the universal file format for call recordings, IVR prompts, voicemail exports, and PSTN debug captures. Every telephony platform (Twilio, Asterisk, FreeSWITCH) reads and writes WAV-wrapped G.711.