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 first 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 Video teleconference and push to talk systems (in conjunction with H.323 or Session Initiation Protocol), 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).

According to RFC 1889, the services provided by RTP include:

RTP ensures consistent delivery order of voice packets in an IP internetwork.

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). RFC 3551 (STD 65) (which obsoletes RFC 1890) defines a specific profile for Audio and Video Conferences with Minimal Control. RFC 3711 defines the SRTP (Secure Real-time Transport Protocol) profile (actually an extension to RTP Profile for Audio and Video Conferences) which can be used (optionally) to provide the confidentiality, message authentication, and replay protection for audio and video streams being delivered.

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