The Real-time Transport Protocol defines a standard packet format for delivering audio and video over the Internet. It is defined in RFC 1889. It was developed by the Audio Video Transport Working group and was first published in 1996. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.
RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number. RTP is one of the foundations of VoIP and it is used in conjunction with SIP which assists in setting up the connections across the network.