RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
The protocol provides facilities for jitter compensation and detection of packet loss and out-of-order delivery, which are common, especially during UDP transmissions on an IP network.
[3] As of 2002, RTP was regarded as the primary standard for audio/video transport in IP networks and is used with an associated profile and payload format.
Real-time multimedia streaming applications require timely delivery of information and often can tolerate some packet loss to achieve this goal.
[5] Other transport protocols specifically designed for multimedia sessions are SCTP[7] and DCCP,[8] although, as of 2012[update], they were not in widespread use.
RTP is used for the transfer of multimedia data, and the RTCP is used to periodically send control information and QoS parameters.
[11] The control protocol, RTCP, is used for quality of service (QoS) feedback and synchronization between the media streams.
To this end, the information required by a specific application of the protocol is not included in the generic RTP header.
[22] The fields in the header are as follows: A functional multimedia application requires other protocols and standards used in conjunction with RTP.
The sender sets the payload type field in accordance with connection negotiation and the RTP profile in use.
It decodes the media data in the packets according to the payload type and presents the stream to its user.