What is rtp
Last updated: April 1, 2026
Key Facts
- RTP (Real-Time Protocol) was developed by the IETF and published in RFC 3550 as the standard for real-time media transport
- RTP runs on top of UDP (User Datagram Protocol) to provide low-latency transmission suitable for live audio and video
- The protocol includes sequence numbers and timestamps to help receivers reconstruct media in correct order and timing
- RTP works alongside RTCP (Real-Time Control Protocol) which monitors delivery quality and provides feedback
- VoIP, video conferencing, streaming services, and live broadcasts commonly rely on RTP for media transmission
Understanding RTP
Real-Time Protocol (RTP) is a standardized network protocol specifically designed for transmitting audio and video over packet-switched networks like the internet. Unlike traditional file transfer protocols that prioritize accuracy and completeness, RTP prioritizes low latency and continuous streaming. It provides the necessary infrastructure for real-time communication by organizing media into packets with timing and sequence information that receivers use to reconstruct the original stream.
How RTP Works
RTP operates by encapsulating audio or video data into packets containing headers with sequence numbers, timestamps, and synchronization source identifiers. These headers allow receiving systems to properly order out-of-sequence packets, synchronize multiple media streams (audio with video), and detect missing packets. RTP typically runs on top of UDP (User Datagram Protocol), which provides faster, lower-latency transmission than TCP at the cost of occasional packet loss.
RTP Components and Features
- Sequence Numbers: Track packet order to detect loss and reorder packets correctly
- Timestamps: Maintain precise synchronization of audio and video streams
- Synchronization Source (SSRC): Identifies the source of each media stream in multi-participant sessions
- Payload Type: Indicates what codec is used (audio, video, type of compression)
- Packet Loss Handling: Allows systems to adapt to missing packets during transmission
RTP and RTCP
RTP works alongside RTCP (Real-Time Control Protocol), a companion protocol that monitors delivery quality and provides feedback to senders. RTCP reports on metrics like packet loss, latency, and jitter to help senders adjust transmission parameters. Together, RTP and RTCP create a complete framework for real-time media delivery with quality monitoring capabilities.
Common RTP Applications
VoIP services, video conferencing platforms like Zoom and Teams, live video streaming, and internet radio all depend on RTP for reliable real-time media delivery. Mobile networks also use RTP for voice and video calls. The protocol's flexibility allows different audio and video codecs to be transported, making it adaptable to various bandwidth conditions and quality requirements.
Related Questions
What is the difference between RTP and TCP?
RTP uses UDP for fast, low-latency transmission suitable for live media, but allows some packet loss. TCP ensures all data arrives correctly but introduces delay. For real-time applications like video calls, RTP's speed advantage outweighs occasional packet loss.
Why does RTP use UDP instead of TCP?
UDP is connectionless and much faster than TCP, which is essential for real-time communication requiring low latency. While UDP doesn't guarantee packet delivery, modern codecs can handle occasional missing packets better than TCP's transmission delays.
What is RTCP and how does it relate to RTP?
RTCP (Real-Time Control Protocol) is a companion protocol that monitors RTP transmission quality and provides feedback. It reports metrics like packet loss and jitter, allowing systems to dynamically adjust streaming quality for optimal user experience.
More What Is in Daily Life
Also in Daily Life
More "What Is" Questions
Trending on WhatAnswers
Browse by Topic
Browse by Question Type
Sources
- Wikipedia - Real-Time Transport ProtocolCC-BY-SA-4.0
- IETF RFC 3550 - RTP SpecificationCC-BY-4.0