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The basic principle behind RTP is very simple: an RTP session comprises a set of participants (we'll also call them peers) communicating with RTP, to either send or receive audio or video. Participants wanting to send will partition the media into different chunks of data called RTP packets, then send those over UDP to the receivers.
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The Real-time Transport Protocol ( RTP ), defined in RFC 3550, is an IETF standard protocol to enable real-time connectivity for exchanging data that needs real-time priority. This article provides an overview of what RTP is and how it functions in the context of WebRTC.
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The Real-time Transport Protocol (RTP) is a network protocol that provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services. In this article: RTP Basics. RTP Packet Header Fields. Payload Type. Sequence Number. Timestamp
Introduction to the Real-time Transport Protocol (RTP)
Real-time Transport Protocol (RTP) is a network standard designed for transmitting audio or video data that is optimized for consistent delivery of live data. It is used in internet telephony, Voice over IP and video telecommunication. It can be used for one-on-one calls (unicast) or in one-to-many conferences (multicast).