RTP – short for 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 an end-to-end transfer of data in real-time. What this means is that data is sent from the server to the client(s) and is in actual time. Imagine something like Internet radio which is audio only, but there should be few lapses in the transmission since the data is usually buffered. A buffer is when data is stored in memory and played from memory. The buffer may allow for a few seconds of stored information before playing. Some applications can allow for setting the buffer amount. 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. The Real-Time Transport Protocol (RTP) is an Internet protocol standard that specifies a way for programs to manage the real-time transmission of multimedia data over either unicast or multicast network services. Originally specified in Internet Engineering Task Force (IETF). RTP was designed by the IETF's Audio-Video Transport Working Group to support video conferences with multiple, geographically dispersed participants. RTP is commonly used in Internet telephony applications. RTP does not in itself guarantee real-time delivery of multimedia data (since this is dependent on network characteristics); it does, however, provide the wherewithal to manage the data as it arrives to best effect. RTP components include: a sequence number, which is used to detect lost packets; payload identification, which describes the specific media encoding so that it can be changed if it has to adapt to a variation in bandwidth; frameindication, which marks the beginning and end of each frame; source identification, which identifies the originator of the frame; and intramedia synchronization, which uses timestamps to detect different delay jitter within a single stream and compensate for it. RTPC components include: quality of service (QoS) feedback, which includes the numbers of lost packets, round-trip time, and jitter, so that the sources can adjust their data rates accordingly; session control, which uses the RTCP BYE packet to allow participants to indicate that they are leaving a session; identification, which includes a participant's name, e-mail address, and telephone number for the information of other participants; and intermedia synchronization, which enables the synchronization of separately transmitted audio and video streams. Compressed RTP (CRTP), specified in RFC 2509, was developed to decrease the size of the IP, UDP, and RTP headers. However, it was designed to work with reliable and fast point-to-point links. In less than optimal circumstances, where there may be long delays, packet loss, and out-of-sequence packets, CRTP doesn't function well for Voice over IP (VoIP) applications. Another adaptation, Enhanced CRPT (ECRPT), was defined in a subsequent Internet Draft document to overcome that problem.

REFERENCE

http://searchnetworking.techtarget.com/definition/Real-Time-Transport-Protocol http://www.3cx.com/pbx/rtp/