Real-Time Carriage Protocol
RTP defines a connected packet architecture for carrying audio and video over the Internet.
It was developed by the Audio-Video Carriage Working Group of the IETF and was first
published in 1996 as RFC 1889, which was fabricated anachronistic in 2003 by RFC 3550.
RTP provides end-to-end arrangement carriage functions advised for applications with realtime
transmission requirements, such as audio and video. Those functions include
payload-type identification, arrangement numbering, time stamping, and commitment monitoring.
Figure 1-6 shows a archetypal role played by RTP in a VoIP network. Specifically, notice
RTP communicates anon amid the articulation endpoints, admitting the alarm bureaucracy protocols
(that is, H.225 and H.245 in this example) are acclimated to acquaint with voice
gateways.
RTP typically runs on top of UDP to use the multiplexing and checksum services of that
protocol. RTP does not have a standard TCP or UDP port on which it communicates. The
only standard it obeys is that UDP communications are done via an even port, and the
next higher odd port is used for RTCP communications. Although no standards are
assigned, in a Cisco environment RTP is generally configured to use UDP ports in the
range 16,384–32,767.
RTP can carry any data with real-time characteristics, such as interactive audio or video.
The fact that RTP uses a dynamic port range can make it difficult for it to traverse
firewalls.
Although RTP is often used for unicast sessions, it is primarily designed for multicast sessions.
In addition to the roles of sender and receiver, RTP defines the roles of translator
and mixer to support multicast requirements.
RTP is frequently used in conjunction with Real-time Streaming Protocol (RTSP) in
streaming media systems. RTP is also used in conjunction with H.323 or SIP in videoconferencing
and push-to-talk systems. These two characteristics make RTP the technical
foundation of the VoIP industry. Applications using RTP are less sensitive to packet loss,
but typically very sensitive to delays, so UDP is a better choice than TCP for such applications.
RTP is a critical component of VoIP because it enables the destination device to reorder
and retime the voice packets before they are played out to the user. An RTP header contains
a time stamp and sequence number, which allow the receiving device to buffer and
to remove jitter by synchronizing the packets to play back a continuous stream of sound.
RTP uses sequence numbers only to order the packets. RTP does not request retransmission
if a packet is lost.