What is the transport protocol in real time?
Real -time transport protocol (RTP) is the standard of the Internet protocol used to perform multimedia Unicast and multicast communications in real time. It consists of two components: a transport protocol and a log in real time (RTCP). The first provides internet protocol specification (IP) for transmission of multimedia flows across network in real time. It provides basic management and quality of service (QoS), such as finding data packet loss and compensation of transmission delay. The real -time transport protocol, which is commonly used in Voice Over Internet Protocol (VOIP), was originally developed in a working group for an internet engineering group working group to provide real -time video conference in RTP. Separate RTP and RTCP packets are transmitted for Each using two different communication ports and/or multiŘečové addresses. Participants may therefore decide to accept only one medium. Synchronized audio and video playback is achieved by using information about RTCP timing for both sound and video sessions.
The Transport Protocol header in real time describes how the codec bit currents are assembled into packets. It also contains instructions that allow you to reconstruct data packets. Other RTP components include the following: Identification of the frame that indicates the start and end of each frame; Intramedia synchronization, which uses time stamps to detect and compensate for shivers delay; and identifying the useful load that describes the media coding method to make adjustments to change the bandwidth.
also part of the transport protocol in real time includes a CE sequence for detecting lost packets and the identification of the source. Components rTCP includes identification that includes participants' names, e -mail addresses, phone numbers, and intermediate synchronization that allow the transfer of separate audio and video streams. The session management allows participants to indicate that they leave the session, while the feedback of the quality of the service (QoS) monitors the number of lost packets; The return and jitter transmission time allows you to adjust the data speed as needed.
Although it provides basic monitoring skills to ensure QoS, RTP does not guarantee the delivery of multimedia communication in real time; Also, RTP does not provide other QoS parameters such as packets in the correct order. It relies on Internet protocols in network and transport layers of the model of open systems (OSI). RTP generally runs on the upper part of the data file (UDP) protocol, although other transport protocols, including the SIP and H.323 initial protocol, can also be used.