A Snapshot of RTP

The key standard for audio/video transport in IP networks is the Real-time Transport Protocol (RTP), along with its associated profiles and payload formats. RTP aims to provide services useful for the transport of real-time media, such as audio and video, over IP networks. These services include timing recovery, loss detection and correction, payload and source identification, reception quality feedback, media synchronization, and membership management. RTP was originally designed for use in multicast conferences, using the lightweight sessions model. Since that time, it has proven useful for a range of other applications: in H.323 video conferencing, webcasting, and TV distribution; and in both wired and cellular telephony. The protocol has been demonstrated to scale from point-to-point use to multicast sessions with thousands of users, and from low-bandwidth cellular telephony applications to the delivery of uncompressed High-Definition Television (HDTV) signals at gigabit rates.

RTP was developed by the Audio/Video Transport working group of the IETF and has since been adopted by the ITU as part of its H.323 series of recommendations, and by various other standards organizations. The first version of RTP was completed in January 1996. ⁶ RTP needs to be profiled for particular uses before it is complete; an initial profile was defined along with the RTP specification, ⁷ and several more profiles are under development. Profiles are accompanied by several payload format specifications, describing the transport of a particular media format. Development of RTP is ongoing, and a revision is nearing completion at the time of this writing. ^{49 , 50}

A detailed introduction to RTP is provided in Chapter 3, The Real-time Transport Protocol, and most of this book discusses the design of systems that use RTP and its various extensions.