Real-Time Transport Protocol (RTP) is an internet protocol used for real-time multimedia communications. It consists of two components: the transport protocol and the real-time transport control protocol (RTCP). RTP is commonly used in VoIP and was developed to conduct real-time videoconferences. RTP and RTCP packets are transmitted separately, and synchronized playback is achieved using timing information in RTCP packets. RTP provides basic monitoring capabilities but does not guarantee real-time delivery or other QoS parameters. It generally runs over UDP but can also use SIP and H.323.
Real-time Transport Protocol (RTP) is an Internet protocol standard used to conduct real-time unicast and multicast multimedia communications. It consists of two components: the transport protocol and the real-time transport control protocol (RTCP). The former provides Internet Protocol (IP) specifications for transmitting media streams across networks in real time. The latter provides basic session management and quality of service (QoS) features, such as packet loss detection and transmission delay compensation. Commonly used in Voice over Internet Protocol (VoIP) telecommunications, Real-Time Transport Protocol was originally developed by the Audio-Video Working Group of the Internet Engineering Task Force to provide a means for conducting real-time videoconferences between multiple participants in locations geographically dispersed.
Audio and video data streams are transmitted separately in RTP. Separate RTP and RTCP packets are each transmitted using two different communication ports and/or multicast addresses. Participants can then choose to receive only one mount. Synchronized playback of audio and video is achieved using timing information in RTCP packets for both audio and video sessions.
The real-time transport protocol header describes how the codec bit streams are assembled into packets. It also contains instructions that allow receiving network devices to reconstruct data packets. Other components of RTP include the following: frame identification, which marks the start and end of each frame; intramedia sync, which uses timestamps to detect and compensate for delay jitter; and payload identification, which describes the method of encoding media so that adjustments can be made for changes in bandwidth.
Also part of the real-time transport protocol are a sequence number for detecting lost packets and an identification of the source. The components of RTCP include identification including participant names, email addresses, telephone numbers, and intermediate synchronization, which allows for the transmission of separate audio and video streams. Session control allows participants to indicate that they are leaving a session while quality of service (QoS) feedback tracks the number of packets lost; the round-trip time and jitter allow the source to adjust the data rate as needed.
While it provides basic monitoring capabilities to ensure QoS, RTP does not guarantee real-time delivery of multimedia communications; nor does RTP ensure other QoS parameters such as packets received in the correct order. To do this, it relies on the Internet protocols in the network and transport layers of the Open Systems Interconnection (OSI) model. RTP generally runs over User Datagram Protocol (UDP), although other transport protocols can also be used, including Session Initiation Protocol (SIP) and H.323.
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