Quality of Service (QoS)
Quality of Service (QoS) for multimedia data transmission depends on many parameters. Some of the most important are:
FIGURE 16.1: Jitters in frame playback: (a) high jitter; (b) low jitter
Multimedia Service ClassesBased on the above measures, multimedia applications can be classified into the following types:
Table Requirement on network bandwidth / bitrate
Perceived QoS. Although QoS is commonly measured by the above technical parameters, QoS itself is a "collective effect of service performances that determine the degree of satisfaction of the user of that service," as defined by the International Telecommunications Union. In other words, it has everything to do with how the user perceives it.
In real - time multimedia, regularity is more important than latency (i.e., jitter and quality fluctuation are more annoying than slightly longer waiting); temporal correctness is more important than the sound and picture quality (i.e., ordering and synchronization of audio and video are of primary importance); and humans tend to focus on one subject at a time.
User focus is usually at the center of the screen, and it takes time to refocus, especially after a scene change. Together with the perceptual nonuniformity we have studied in previous chapters, many issues of perception can be exploited in achieving the best perceived QoS in networked multimedia.
TableTolerance of latency and jitter in digital audio and video
QoS for IP Protocols
QoS policies and technologies enable key metrics discussed in the previous section such as latency, packet loss, and jitter to be controlled by offering different levels of service to different packet streams or applications.
Frame relay routing protocol and ATM provide some levels of QoS, but currently most Internet applications are built on IP. IP is a "best - effort" communications technology and does not differentiate among different IP applications. Therefore it is hard to provide QoS over IP by current routing methods.
Abundant bandwidth improves QoS, but in complex networks, abundant bandwidth is unlikely to be available everywhere (in practice, many IP networks routinely use oversubscription). In particular, it is unlikely to be available in all the access links. Even if it is available everywhere, bandwidth alone can't resolve problems due to sudden peaks in traffic.
Differentiated Service (DiffSety) uses DiffServ code [Type of Service (TOS) octet in IPv4 packet and Traffic Class octet in IPv6 packet] to classify packets to enable their differentiated treatment, It is becoming more widely deployed in intradomain networks and enterprise networks, as it is simpler and scales well, although it is also applicable to end - to - end networks. DiffServ, in conjunction with other QoS techniques, is emerging as the de facto QoS technology. See IETF Request for Comments (RFC) 2998 for more information.
Multiple Protocol Label Switching (MPLS) facilitates the marriage of IP to OSI layer 2 technologies, such as ATM, by overlaying a protocol on top of IP. It introduces a 32 - bit label and inserts one or more shim labels into the header of an IP packet in a backbone IP network. It thus creates tunnels, called Label Switched Paths (LSP). By doing so, the backbone IP network becomes connection - oriented.
The two main advantages of MPLS are to support Traffic Engineering (TE), which is used essentially to control traffic flow, and Virtual Private Networks (VPN). Both TE and VPN help delivery of QoS for multimedia data. MPLS supports eight service classes. For more detail refer to RFC 3031.
DiffServ and MPLS can be used together to allow better control of both QoS performance per class and provision of bandwidth, retaining advantages of both MPLS and DiffServ.
When a high packet loss or error rate is detected in the event of network congestion, prioritized delivery of multimedia data can be used to alleviate the perceived deterioration.
Prioritization for types of media. Transmission algorithms can provide prioritized delivery to different media — for example, giving higher priority to audio than to video — since loss of content in audio is often more noticeable than in video.
Prioritization for uncompressed audio. PCM audio bitstreams can be broken into groups of every nth sample — prioritize and send k of the total of n groups (k ≤ n) and ask the receiver to interpolate the lost groups if so desired. For example, if two out of four groups are lost, the effective sampling rate is 22.05 kHz instead of 44.1 kHz. Loss is perceived as change in sampling rate, not dropouts.
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Introduction To Multimedia
Multimedia Authoring And Tools
Graphics And Image Data Representations
Colour In Image And Video
Fundamental Concepts In Video
Basics Of Digital Audio
Lossless Compression Algorithm
Lossy Compression Algorithms
Image Compression Standards
Basic Video Compression Techniques
Mpeg Video Coding I – Mpeg 1 And 2
Mpeg Video Coding Ii- Mpeg-4, 7, And Beyon
Basic Audio Compression Techniques
Mpeg Audio Compression
Computer And Multimedia Networks
Multimedia Network Communications And Applications
Content-based Retrieval In Digital Libraries
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