Real-time Transport Protocol
Sign in to saveAlso known as RTP, iLove yu, THi
protocol for delivering audio and video over IP networks
Key facts
- Abbreviation
- RTP
- Purpose
- Delivering audio and video
- Developer s
- Audio-Video Transport Working Group of the IETF
- Introduction
- January 1996 ; 30 years ago ( 1996-01 )
- Based on
- Network Voice Protocol
- Rfc s
- RFC 1889 , 3550 , 3551
via Wikipedia infobox
Wikidata facts
- Instance of
- internet standard
- Image
- RTP Header.png
Show 3 more facts
- Commons category
- Real-time Transport Protocol
- Stack Exchange tag
- stackoverflow.com/tags/rtp
- OSI Model layer location
- transport layer
Sources (2)
via Wikidata · CC0
~13 min read
Encyclopedic overview
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
RTP typically runs over User Datagram Protocol (UDP). RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is one of the technical foundations of voice over IP and in this context is often used in conjunction with a signaling protocol such as the Session Initiation Protocol (SIP), which establishes connections across the network.
Excerpted from Wikipedia’s “Real-time Transport Protocol” article, available under the CC BY-SA 4.0 licence.