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Netw 320 Homework Wk 6

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Submitted By tsouthern
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Real-Time Transport Protocol
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NETW 320
10/3/2014

Real-Time Transport Protocol (RTP) is a way for programs to manage the real-time multimedia, such as, audio / video, data over unicast (point to point) and/or multicast (multipoint) network services. RTP is used extensively in systems that involve streaming. VoIP, video teleconference applications and several television services utilize RTP. RTP provides an IP-based transmission service for real-time communication between computers or devices. In combination with other protocols, RTP is used for transmitting audio and video information. VoIP uses a combination of protocols to transmit the phone data between computers or from a computer to a cell phone or land-line. VoIP data is changed from an analog signal to a digital signal. Also RTP allows discontinuous transmission (silence suppression) on any audio payload format. The receiver can detect silence suppression on the first packet received after the silence by observing that the RTP timestamp is not contiguous with the end of the interval covered by the previous packet even though the RTP sequence number has incremented only by one ( voipinfo, 2003). Layered protocols transfer, receive and interpret the signal; RTP is a part of the transport layer. RTP itself does not provide a mechanism to secure the transmission quality. It is however, able to ensure real-time communication even without such mechanisms. It uses consecutive numbering of the sent data packets. Based on this numbering the recipient can rearrange information that arrived in the wrong order. Numbering also serves as a way to determine the correct position of a data fragment within a data stream. This is particularly important for audio and video transmissions, from which information is continuously lost due to poor transmission quality. RTP works in conjunction with the User Datagram Protocol (UDP) to transfer the messages. The UDP protocol data is delivered faster, but loses reliability of delivery. In most instances, a dropped frame should not affect playback too badly.
Playback is only affected if the network is severely congested causing slow delivery. It is possible to use RTP over the Transmission Control Protocol (TCP) since it is reliable but it often has problems with delivering data in a timely fashion. RTP and its companion RTCP (Real time Transport Control Protocol) make it possible to respectively transport and monitor data blocks which have real time properties. RTP and RTCP are protocols which are located at application level and use underlying TCP or UDP transport protocols. RTCP is a control protocol associated with RTP; it measures performances but offers no guarantees. Doing so, it must use a reservation protocol such as RSVP or make sure that the communication links are correctly proportioned in relation to the use which is made of it. There are several RFC’s associated with RTP. Some of them are as follows: * RFC 3350 (A Real-Time Protocol Applications), * RFC 3551(RTP Profile for Audio and Video Conferences with Minimal Control), * RFC 4585 (Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF), * RFC 5117 (RTP Topologies), * RFC 5761 (Multiplexing RTP Data and Control Packets on a Single Port) * RFC 2250(RTP Payload Format for MPEG1/MPEG2 Video)

References

(n.d.). Retrieved from http://www.ietf.org/
What is real-time transport protocol (rtp). (2000). Retrieved from http://searchnetworking.techtarget.com/definition/Real-Time-Transport-Protocol
Voip-info.org (2014) http://www.voip-info.org/wiki/view/RTP+Silence+Suppression

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