THE WHATIS.COM WORD-OF-THE-DAY   
March 25, 2002

Real-Time Transport Protocol 
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TODAY'S WORD: Real-Time Transport Protocol 

See our definition with hyperlinks at
http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci812019,00.html 

The Real-Time Transport Protocol (RTP) is an Internet protocol
standard that specifies a way for programs to manage the real-time
transmission of multimedia data over either unicast or multicast
network services. Originally specified in Internet Engineering Task
Force (IETF) Request for Comments (RFC) 1889, RTP was designed by the
IETF's Audio-Video Transport Working Group to support video
conferences with multiple, geographically dispersed participants. RTP
is commonly used in Internet telephony applications. RTP does not in
itself guarantee real-time delivery of multimedia data (since this is
dependent on network characteristics); it does, however, provide the
wherewithal to manage the data as it arrives to best effect. 

RTP combines its data transport with a control protocol (RTCP), which
makes it possible to monitor data delivery for large multicast
networks. Monitoring allows the receiver to detect if there is any
packet loss and to compensate for any delay jitter. Both protocols
work independently of the underlying Transport layer and Network
layer protocols. Information in the RTP header tells the receiver how
to reconstruct the data and describes how the codec bit streams are
packetized. As a rule, RTP runs on top of the User Datagram Protocol
(UDP), although it can use other transport protocols. Both the
Session Initiation Protocol (SIP) and H.323 use RTP. 

RTP components include: a sequence number, which is used to detect
lost packets; payload identification, which describes the specific
media encoding so that it can be changed if it has to adapt to a
variation in bandwidth; frame indication, which marks the beginning
and end of each frame; source identification, which identifies the
originator of the frame; and intramedia synchronization, which uses
timestamps to detect different delay jitter within a single stream
and compensate for it. 

RTPC components include: quality of service (QoS) feedback, which
includes the numbers of lost packets, round-trip time, and jitter, so
that the sources can adjust their data rates accordingly; session
control, which uses the RTCP BYE packet to allow participants to
indicate that they are leaving a session; identification, which
includes a participant's name, e-mail address, and telephone number
for the information of other participants; and intermedia
synchronization, which enables the synchronization of separately
transmitted audio and video streams. 

Compressed RTP (CRTP), specified in RFC 2509, was developed to
decrease the size of the IP, UDP, and RTP headers. However, it was
designed to work with reliable and fast point-to-point links. In less
than optimal circumstances, where there may be long delays, packet
loss, and out-of-sequence packets, CRTP doesn't function well for
Voice over IP (VoIP) applications. Another adaptation, Enhanced CRPT
(ECRPT), was defined in a subsequent Internet Draft document to
overcome that problem.

RELATED TERMS:

real time
http://searchsolaris.techtarget.com/sDefinition/0,,sid12_gci214344,00.html

unicast 
http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci213249,00.html 

multicast 
http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci212610,00.html 

User Datagram Protocol 
http://searchwebmanagement.techtarget.com/sDefinition/0,,sid27_gci214157,00.html 

Voice over IP 
http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci214148,00.html 
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SELECTED LINKS:

The IETF provides the Real-Time Transport Protocol RFC. 
http://www.ietf.org/rfc/rfc1889.txt 

The Audio/Video Transport Working Group maintains an RTP
Frequently-Asked Questions list. 
http://www.cs.columbia.edu/~hgs/rtp/faq.html 

SearchNetworking offers "Best Web Links about Voice/Data
Convergence." 
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