RTP

RTP (Real-time Transport Protocol) has been designed for the transport of multimedia streams over IP networks. It works on top of UDP. It was developed by the Audio-Video Transport Working Group of the IETF[1] and first published in 1996 as RFC 1889, and superseded by RFC 3550 in 2003.

IP networks are asynchronous and may lose or reorder packets.

Some fields in RTP are defined to address these problems:

- PT (Payload Type) tags the content of the packet. Some standardized values exist, but for other codecs, the type must be described in a service description protocol, SDP (Session DescriptionProtocol);

- sequence number is primarily used to identify and detect lost packets and secondly to reconstruct the order in which packets where sent, which may make loss detection easier;

- the timestamp is the sampling instant for the first octet of media data in a packet. It can be used to help recover the clock frequency at the receiving side, if it is not given by other means.

RTP does not include a loss recovery mechanism in case of packet losses.