The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and published in 1996 as RFC 1889.

It was originally designed as a multicast protocol, but has since been applied in many unicast applications. It is frequently used in streaming media systems (in conjunction with RTSP) as well as videoconferencing and push to talk systems (in conjunction with H.323 or SIP), making it the technical foundation of the Voice over IP industry. It goes along with the RTP Control Protocol (RTCP) and it's built on top of User Datagram Protocol (in OSI model).

RTP was also published by the ITU-T as H.225.0, but later removed once the IETF had a stable standards-track RFC published. It exists as an Internet Standard (STD 64) defined in RFC 3550 (which obsoletes RFC 1889). RFC3551 (STD 65) (which obsoletes RFC 1890) defines a specific profile for Audio and Video Conferences with Minimal Control.

References

• Perkins, Colin (2003). RTP: Audio and Video for the Internet (1st ed.) Addison-Wesley Pub Co. ISBN 0672322498

External link

• Henning Schulzrinne's RTP page.