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Glossary · Voice technology

What is RTP (real-time transport protocol)?

  • Also called: real-time transport protocol, rtp stream, media stream

Reviewed by Sujan ThapaliyaLast updated

Definition

RTP carries the actual audio of a VoIP call, once SIP has set the call up. It runs over UDP, adds timestamps and sequence numbers so packets can be reassembled in order, and accepts losing a little data rather than waiting for retransmission.
The three STIR/SHAKEN attestation levels, which underpin how rtp (real-time transport protocol) behaves in practice: A asserts both that the carrier knows the customer and that the customer may use the calling number, B asserts the customer only, and C asserts neither. A shared number pool can reach B at best, which is why number ownership keeps recurring in these definitions.
Attestation is the backdrop to most caller-ID and deliverability terms, rtp (real-time transport protocol) included.

In practice

How rtp (real-time transport protocol) actually works

That trade-off is why voice sounds different from a download. A late packet is useless to a conversation, so RTP discards it rather than asking for it again, which is what makes packet loss audible rather than merely slow.

Media travels directly between endpoints where the network allows, which is efficient and also why NAT and firewalls cause one-way audio: signalling completed, but the media path did not.

Encrypted RTP — SRTP — protects the audio in transit and is what a call described as end-to-end encrypted is generally using.

RTP also carries the timing information that makes recording and transcription possible after the fact. Packets arriving out of order are reassembled by sequence number, so a recording can be perfect even when the live call sounded broken — which is why a customer's complaint about audio quality often cannot be reproduced from the recording.

Worked example

A 20-millisecond gap in RTP is one lost packet, and concealment usually hides it. Twenty such gaps a minute is the choppy audio people describe as the line breaking up.

RTP (real-time transport protocol): common questions

Why does RTP use UDP rather than TCP?
Because retransmitting a lost packet takes longer than the conversation can wait. Voice tolerates small losses far better than it tolerates delay, so RTP prioritises timing over completeness.
What causes one-way audio?
A media path that only works in one direction, usually NAT or firewall configuration. Signalling completes because it takes a different route, which is why the call connects before it fails.
Is RTP encrypted?
Not by default. SRTP is the encrypted variant, and it has to be negotiated by both ends. Signalling encryption over TLS and media encryption over SRTP are separate settings.

Sources

  1. Combating Spoofed Robocalls with Caller ID Authentication — Federal Communications CommissionThe STIR/SHAKEN framework, the attestation levels carriers sign calls with, and the mandate requiring providers to authenticate caller ID.
  2. 47 U.S.C. § 227 — Restrictions on the use of telephone equipment — Cornell Legal Information InstituteThe Telephone Consumer Protection Act itself — the consent requirements, calling-hours limits, and private right of action.
  3. ITU-T Recommendation E.164 — The international public telecommunication numbering plan — International Telecommunication UnionThe international number format, the 15-digit maximum, and how country codes and national numbers compose.

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