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Real-time Transport Protocol (RTP)

Real-time Transport Protocol (RTP)

RTP (version 2) is a real-time transport protocol that provides end-to-end delivery services to support applications transmitting real-time audio and video, over unicast and multicast network services.

RTP is defined in IETF RFC 3550, along with a profile for carrying audio and video over RTP in RFC 3551.

RTP provides end-to-end delivery services, but it does not provide all the functionality that is typically provided by a transport protocol. In fact, RTP typically runs on top of UDP to utilise its multiplexing and checksum services. Other transport protocols besides UDP can carry RTP as well.

RTP does not:

  • Provide any mechanism to guarantee quality of service, but relies on lower-layer services to do so.
  • Guarantee delivery or prevent out-of-order delivery.

In terms of VoIP, you could think of RTP as looking after the transmission of the call, while SIP sets up the call at the start and closes it at the other end. RTP is used for more than just VoIP, though. It is a big part of online services like WhatsApp, FaceTime, and Teams and can be used anywhere where live video and audio streaming is needed.

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In this Course, you’ll further explore the web protocols that underpin the internet and the world wide web, and some of the applications they enable.

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