 carried in the payload of the RTP packet. It is used for playing back media at the correct speed, and together with RTCP, it is used for syn- chronizing the presentation of other streaming media. Applications typically run RTP on top of UDP to make use of its multiplexing and checksum services. Both protocols con- tribute parts of the transport protocol functionality. RTP supports data transfer to multiple destinations using multicast distribution if provided by the underlying network. RTP does not provide any mechanism to ensure timely delivery or provide other QoS guarantees, but relies on lower layer services to do SO. It does not guarantee delivery or prevent out-of-order delivery, nor does it assume that the underlying network is reliable and delivers packets in sequence. The sequence numbers included in RTP allow the receiver to reconstruct the sender's packet sequence, but sequence numbers might also be used to determine the proper location of a packet, for example in video decoding, without necessarily decoding packets in sequence. RTCP is the control protocol that works in conjunction with RTP. RTCP control packets are periodically transmitted by each participant in an RTP session to all other participants. The primary function is to provide feedback on the quality of the data distribution. The feedback may be directly useful for control of adaptive encodings but experiments with IP multicasting have shown that it is also critical to get feedback from the receivers to diagnose faults in the distribution. Sending reception feedback reports to all participants allows determining whether possible problems are local or global. RTSP (RFC 2326) The real-time streaming protocol (RTSP) is used as a session control protocol for media streaming applications. Several features and semantics in RTSP have been inherited from HTTP. Using RTSP, a client streaming application can establish a session with a media streaming server. Using this session, the client can ask the server: to start streaming media; to pause, back