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The Real-Time Transport Protocol (RTP) is a protocol for the continuous transmission of audiovisual data (streams) via IP-based networks. The protocol was first standardized in RFC 1889 in 1996. In 2003 it was replaced by RFC 3550.

It is used to transport multimedia data streams (e.g. audio data in Internet telephony) over networks, i.e. to encode, pack and send the data.  RTP is a packet-based protocol and is usually operated via UDP.  RTP can be used for both unicast connections and multicast communication on the Internet. The RealTime Control Protocol (RTCP) works with RTP to negotiate and maintain Quality of Service (QoS) parameters.

SRTP is ideal for VoIP traffic protection because it can be used in conjunction with header compression and does not affect the quality of the IP service. This brings decisive advantages especially for data traffic that uses voice codecs with low transmission rates.

The main function of  RTP is the transmission of data streams that require real-time, while the Real-Time Streaming Protocol (RTSP) is used to control and monitor data transmission.

The Datagram Congestion Control Protocol (DCCP) is a current approach to enable congestion control for RTP/UDP-based media flows.