Real-Time Transport Protocol (RTP) is an Internet Protocol standard that specifies the way programs manage the real-time transmission of multimedia data over unicast or multicast network services.

In comparison to TCP (Transmission Control Protocol) which favors data integrity rather than delivery speed, RTP favors rapid delivery and has mechanisms to compensate for any minor loss of data integrity.

RTP defines the standardized packet format for delivering audio and video over IP networks and used in conjunction with Real-Time Transport Control Protocol (RTCP) to ensure that multiple streams of media can be synchronized and Quality of Service (QoS) can be maintained.

In distributed computing, losing information can be catastrophic but in media streaming, packet losses can be catered for through clever algorithms that patch undelivered/late packet data in real time. Up to a point - there are obviously Quality of Service limits that are deemed acceptable/unacceptable. RTP allows the opportunity to apply frame padding to prevent viusal stuttering and to apply algorithms that patch audio dropouts and prevent clicks or extraneous digital noise.

The most important recent application of RTP is the introduction of VoIP (Voice over Internet Protocol) systems which are becoming very popular as alternatives to regular telephony circuits.