RTP

showstack / protocols / rtp

Also called Real-time Transport Protocol, RTP/RTCP.

The framing that carries real-time media over UDP — sequence numbers, a timestamp and a payload type on every packet, so a receiver can reorder, detect loss and reconstruct timing. AES67, ST 2110 and most VoIP intercom are RTP underneath.

Network transportOpenIETFReported

Media Large, continuous, on time and in order

Continuous streams that are large enough to be sized properly, and that must arrive in order and on time.

What kills it. Anything that starves the link — usually a management transfer nobody scheduled — and jitter that outruns the receive buffer.

Why this one, and not what the category suggests. The transport AES67, RAVENNA and ST 2110 all actually run on.

The four flows, and why the model is worth having →

What goes wrong

RTP carries no reliability of its own. There are no retransmissions, so a lost packet is a lost packet — which is the correct trade for live media and the wrong one for file transfer.

The timestamp is a media clock, not a wall clock. Aligning several RTP streams to each other needs a shared reference such as PTP.

Port numbers are assigned by the signalling that set the session up, not by RTP itself, which is why a firewall rule for "RTP" is rarely a single port.

Typical use

The media layer under AES67 and SMPTE ST 2110
VoIP and IP intercom
Streaming contribution feeds alongside RTSP or SIP signalling

Sources

RFC 3550 — RTP, A Transport Protocol for Real-Time Applications — IETF primary

Understand the mechanism

Show networks

Something wrong, or missing?

Known gaps on this entry: default_ports, implementations. If you can source one of them, that is a ten minute pull request.

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