Performance Analysis of Multimedia Traffic

Performance Analysis of Multimedia Traffic

Federico Montesino Pouzols (University of Seville, Spain), Angel Barriga Barros (University of Seville, Spain), Diego R. Lopez (RedIRIS, Spain) and Santiago Sánchez-Solano (CSIC - Scientific Research Council, Spain)
Copyright: © 2008 |Pages: 6
DOI: 10.4018/978-1-59904-885-7.ch158
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Videoconferencing and multimedia communication technologies in general are gaining momentum with the development of networked organizations. Efforts are currently underway to further adapt multimedia technologies to virtual organizations where multimedia communications find applications in training and education, customer relationships management, enterprise resource planning and many other areas (Camarinha-Matos, 2002).

Key Terms in this Chapter

Forward Error Correction (FEC): A method for performing error control in data transmission in which the source sends redundant data and the destination recognizes only the portion of the data that contains no apparent errors. FEC differs from standard error detection and correction in that the technique is specifically designed to allow the receiver to correct some errors without having to request a retransmission of data. It is thus specially suited for interactive or delay sensitive multimedia networked applications. FEC codes impose a greater bandwidth overhead than backward error correction protocols, but are able to recover from errors more quickly and with significantly fewer re-transmissions.

Voice Over IP (VoIP): The routing of voice communications over the Internet or any other IP-based network. The voice data flows over a general-purpose packet-switched network, instead of traditional dedicated, circuit-switched telephony transmission lines.

Codec (Coder-Decoder): A device or technology for compressing and decompressing data that can be implemented in software, hardware or a combination of both. Often used in videoconferencing and streaming applications, codecs encode a stream or signal for transmission, storage or encryption and decode it for viewing or editing. Speech codecs are designed to deal with the characteristics of voice, while audio codecs are developed for music.

Videoconferencing: A conference between two or more participants at different sites by using a computer network to transmit audio, video and other media. A point-to-point videoconferencing system works like a video telephone. Multipoint videoconferencing allows three or more participants to sit in a virtual conference room.

Internet Telephony (IP Telephony): The fusion of VoIP (packet-switched) telephony systems and traditional circuit-switched telephony systems.

Session Initiation Protocol (SIP): a standard from the Internet Engineering Task Force (IETF) for initiating, modifying and terminating interactive user sessions that involve multimedia communications, such as audio, video, instant messaging, online games, simulation environments and virtual reality. SIP is a peer-to-peer protocol. Thus, it requires only a very simple (and highly scalable) core network with intelligence distributed to the network edge, embedded in endpoints or terminals.

Content Delivery Network (CDN): A computer network system in which nodes cooperate transparently to deliver multimedia content to end users. A number of architectures for implementing CDN systems have been proposed for facilitating replication for content providers, improving scalability and end-user perceived performance and reducing bandwidth usage.

H.323: A set of recommendations from the ITU-T that defines the protocols to provide audio-visual communication sessions on any packet network. H.323 is a part of the H.32x series of protocols which also address communications over ISDN, PSTN and SS7. One strength of H.323 was the relatively early availability of a set of standards, not only defining the basic call model, but in addition the supplementary services needed to address business communication expectations.

Streaming: Streaming media is a multimedia delivery system in which media is consumed while it is being delivered. Widespread deployment of streaming media raises scalability and quality-of-service issues.

Real-Time Transport Protocol RTP: The standard real-time transport protocol from the Internet Engineering Task Force (IETF). RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. The data transport is augmented by a companion control protocol (RTCP) to allow monitoring of the data delivery. RTP is used for transport of multimedia flows within both SIP and H.323 applications.

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