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Bit Rate (bit + rate)
Selected AbstractsSmoothing and transporting video in QoS IP networksINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 7 2006Khaled Shuaib Abstract Real-time traffic such as voice and video, when transported over the Internet, demand stringent quality of service (QoS) requirements. The current Internet as of today is still used as a best effort environment with no quality guarantees. An IP-based Internet that supports different QoS requirements for different applications has been evolving for the past few years. Video streams are bursty in nature due to the instant variability of the video content being encoded. To help mitigate the transport of bursty video streams with minimal loss of information, rate-adaptive shapers (RASs) are usually being used to reduce the burstiness and therefore help preserve the desired quality. When transporting video over a QoS IP network, each stream is classified under a specific traffic profile to which it must conform, to avoid packet loss and picture quality degradation. In this paper we study, evaluate and propose RASs for the transport of video over a QoS IP network. We utilize the encoding video parameters for choosing the appropriate configuration needed to support the real-time transport of Variable Bit Rate (VBR) encoded video streams. The performance evaluation of the different RASs is based on the transport of MPEG-4 video streams encoded as VBR. The performance is studied based on looking at the effect of various parameters associated with the RASs on the effectiveness of smoothing out the burstiness of video and minimizing the probability of packet loss. Copyright © 2005 John Wiley & Sons, Ltd. [source] Novel DLC model for QoS enhancement of bursty VBR traffic in wireless ATM networksEUROPEAN TRANSACTIONS ON TELECOMMUNICATIONS, Issue 8 2008Hosam El-Ocla Several data-link control (DLC) protocol procedures have been proposed in order to provide reliable data transmission over powerless radio links. However, many quality of service (QoS) issues still need to be achieved such as balance between cell transfer delay (CTD) and cell loss rate (CLR), absence of cell delay variation (CDV) and network traffic utilisation. The main problem with wireless ATM is how to overcome the unreliability of the wireless link in order to maintain QoS requirements especially for variable bit rate (VBR) application. Here, we propose a model that provides QoS support for ATM virtual connections (VC) and solves the unreliability problem of the wireless ATM through an efficient DLC protocol for bursty VBR traffic. This model mainly concerns with enhancing QoS to gain error-free wireless transmission and regulating VBR Traffic. Copyright © 2008 John Wiley & Sons, Ltd. [source] Experimental analysis of the impact of peer-to-peer applications on traffic in commercial IP networksEUROPEAN TRANSACTIONS ON TELECOMMUNICATIONS, Issue 6 2004Nadia Ben Azzouna To evaluate the impact of peer-to-peer (P2P) applications on traffic in wide area networks, we analyze measurements from a high speed IP backbone link carrying TCP traffic towards several ADSL areas. The first observations are that the prevalent part of traffic is due to P2P applications (almost 80% of total traffic) and that the usage of network becomes symmetric in the sense that customers are not only clients but also servers. This latter point is observed by the significant proportion of long flows mainly composed of ACK segments. When analyzing the bit rate created by long flows, it turns out that the TCP connections due to P2P applications have a rather small bit rate and that there is no evidence for long range dependence. These facts are intimately related to the way P2P protocols are running. We separately analyze signaling traffic and data traffic. It turns out that by adopting a suitable level of aggregation, global traffic can be described by means of usual tele-traffic models based on M/G/, queues with Weibullian service times. Copyright © 2004 AEI [source] On multi-cell admission control in CDMA networksINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 1 2008Gábor Fodor Abstract We consider a multi-cell (MC) code division multiple access (CDMA) system that supports multiple service classes, including peak rate allocated and elastic ones. Peak rate allocated sessions,when admitted into the system,transmit at a constant bit rate, while elastic sessions can be slowed down at the expense of increasing their residency time. Admitted sessions cause an instantaneous bit rate-dependent interference in neighbour cells. In this rather general setting, we propose a method to calculate the class-wise blocking probabilities as the functions of the estimated so-called inter-cell coupling factors. In the paper this coupling factor is the ratio between the uplink path gains to different Node-B:s (that can be easily obtained in a CDMA system from pilot measurement reports), but our model could include other coupling measures as well. We find that when these coupling factors are underestimated, the system may get into false states (FSs) or false rate states (FRSs) that lead to violating the noise rise threshold. As traffic becomes increasingly elastic, the probability of FSs decreases, but the probability of FRSs increases. Based on numerical results, we make the point that as the traffic becomes more elastic, avoiding the underestimation of these coupling factors as well as exercising MC admission control plays an increasingly important role in guaranteeing proper service quality. Copyright © 2007 John Wiley & Sons, Ltd. [source] Performance analysis and interoperability issue of consolidation algorithms in point-to-multipoint ABR servicesINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 1 2002Naris Rangsinoppamas Abstract In this paper we propose a performance analysis of the consolidation algorithms in point-to-multipoint ABR services in ATM network. We also present a new efficient consolidation algorithm, which overcomes some drawbacks of the previous works. The performance of the previous works and our proposed consolidation algorithms are analysed mathematically. General equations for calculating the response time and allowed cell rate of the source are derived. The derived equations show relatively good agreement with simulation results and can be applied for using in various network topologies. In addition, the interoperability issue in multicast available bit rate (ABR) is investigated. It is found that the branch points implemented with different consolidation algorithms can work interoperably. It is concluded from the simulation results that the performance of the point-to-multipoint ABR services is dominated by the most upper stream branch point (the nearest one to the source), therefore, in order to avoid the consolidation noise and get a faster response in the network, the most upper stream branch point has to be implemented with a high accuracy consolidation algorithm while the lower stream branch point(s) should be implemented with a fast response consolidation algorithm with a less degree of complexity. Copyright © 2002 John Wiley & Sons, Ltd. [source] A speech coder for PC multimedia net-to-net communicationINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 7 2001Shahid Parvez Abstract The paper presents a speech coding algorithm for operation at 11025 samples/s. The coder provides improved speech quality and compatibility with the MS-Windows multimedia environment. The coding algorithm has been developed by adapting the ITU G729 and enhancing it with some recent developments in the medium band coding. The coder operates over a band of frequencies ranging from 20 to 5400 Hz at a bit rate of 8.9 kbit/s. Application of this coder includes intranet VoIP, voice chatting, multimedia communications, and voice archiving. Copyright © 2001 John Wiley & Sons, Ltd. [source] A simple, scalable and provably stable explicit rate computation scheme for flow control in communication networksINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 6 2001James Aweya Abstract This paper describes fast rate computation (FASTRAC), an explicit rate flow control algorithm for available bit rate (ABR) traffic. Using digital control theory, we develop a simple rate controller for the ABR flow control process. We prove that the controller is stable, fair to all participating sources and configurable with respect to responsiveness. The analysis presented shows that stability of the flow control process depends primarily on two factors, the control update rate and the feedback delay. The implementation of the proposed algorithm is much simpler than other fair rate allocation algorithms. The proposed algorithm demonstrates the ability to scale with speed, distance, different feedback delays, number of users, and number of nodes while remaining robust, efficient, and fair under stressing and dynamic traffic conditions. Copyright © 2001 John Wiley & Sons, Ltd. [source] Subband decomposition strategies for object-based image codingINTERNATIONAL JOURNAL OF IMAGING SYSTEMS AND TECHNOLOGY, Issue 3 2003Marcio Albuquerque de Souza Abstract In order to attain perfect reconstruction, subband decomposition of arbitrarily shaped objects requires special strategies to deal with the decimation problem. One of the two strategies described in this paper (scheme 1) inserts pixels with amplitudes equal to the mean values of the neighboring ones belonging to the object. The requirement of transmitting (coding) these pixels reduces the coding gain and is avoided by the second strategy (scheme 2) examined in this paper. This strategy determines specific values of the pixels added to the object in such a way that they result in predefined amplitudes after decimation. For this reason, this method outperforms scheme 1. PSNR versus bit rate are presented for two objects. © 2003 Wiley Periodicals, Inc. Int J Imaging Syst Technol 13, 179,187, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ima.10054 [source] Performance comparisons for adaptive LEO satellite linksINTERNATIONAL JOURNAL OF SATELLITE COMMUNICATIONS AND NETWORKING, Issue 3 2006William G. Cowley Abstract This paper considers the potential to achieve improved throughput in time-varying satellite links which have flexibility in information bit rate and/or transmit power. We assume that other parameters of the link budget such as antenna gains and operating frequency are fixed. Simple results are derived, which illustrate what improvements in data throughput or power consumption are possible under two low-earth orbit scenarios: inter-satellite links and satellite to ground communications. Copyright © 2006 John Wiley & Sons, Ltd. [source] Simulation analyses of weighted fair bandwidth-on-demand (WFBoD) process for broadband multimedia geostationary satellite systemsINTERNATIONAL JOURNAL OF SATELLITE COMMUNICATIONS AND NETWORKING, Issue 4 2005Güray Açar Abstract Advanced resource management schemes are required for broadband multimedia satellite networks to provide efficient and fair resource allocation while delivering guaranteed quality of service (QoS) to a potentially very large number of users. Such resource management schemes must provide well-defined service segregation to the different traffic flows of the satellite network, and they must be integrated with some connection admission control (CAC) process at least for the flows requiring QoS guarantees. Weighted fair bandwidth-on-demand (WFBoD) is a resource management process for broadband multimedia geostationary (GEO) satellite systems that provides fair and efficient resource allocation coupled with a well-defined MAC-level QoS framework (compatible with ATM and IP QoS frameworks) and a multi-level service segregation to a large number of users with diverse characteristics. WFBoD is also integrated with the CAC process. In this paper, we analyse via extensive simulations the WFBoD process in a bent-pipe satellite network. Our results show that WFBoD can be used to provide guaranteed QoS for both non-real-time and real-time variable bit rate (VBR) flows. Our results also show how to choose the main parameters of the WFBoD process depending on the system parameters and on the traffic characteristics of the flows. Copyright © 2005 John Wiley & Sons, Ltd. [source] Visible light communication systems for optical video transmissionMICROWAVE AND OPTICAL TECHNOLOGY LETTERS, Issue 7 2010J. Rufo Abstract This article describes an optical wireless system based on visible light communication (VLC), which allows a video broadcasting to reach a bit rate of 1 Mbps, although this system can also be used for low-speed sensor interconnections. The main advantages of this technology are the robustness against EM interference, safety for human eye, and security against undesired network access. In this article, we present the electronic structure of a low-cost VLC transceiver, the modulation process and the Ethernet interface that has been implemented in each AP (access point). Finally, some conclusions are drawn. © 2010 Wiley Periodicals, Inc. Microwave Opt Technol Lett 52: 1572,1576, 2010; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/mop.25236 [source] Harmonic broadcasting is bandwidth-optimal assuming constant bit rateNETWORKS: AN INTERNATIONAL JOURNAL, Issue 3 2006Lars Engebretsen Abstract Harmonic broadcasting was introduced by Juhn and Tseng in 1997 as a way to reduce the bandwidth requirements required for video-on-demand broadcasting. In this article, we note that harmonic broadcasting is actually a special case of the priority encoded transmission scheme introduced by Albanese et al. in 1996, and prove,using an information theoretic argument,that it is impossible to achieve the design goals of harmonic broadcasting using a shorter encoding. © 2006 Wiley Periodicals, Inc. NETWORKS, Vol. 47(3), 172,177 2006 [source] A new progressive coding scheme for halftone dithered images using bit-interleavingINTERNATIONAL JOURNAL OF IMAGING SYSTEMS AND TECHNOLOGY, Issue 4 2006Jing-Ming Guo Abstract A novel progressive coding scheme is presented for the efficient display of dithered images. Dithered images are the results of thresholding original gray-level images with dithering screens. After the preprocessing of bit-interleaving, this algorithm utilizes the characteristic of reordered image to determine the transmitting order and then progressively reconstructs the dithered image. In addition, the dithered images are further compressed by lossy and lossless procedures. The experimental results demonstrate high-quality reconstructions while maintaining low transmitted bit rates. Moreover, an objective error criterion obtained by LMS optimization is also presented in this study. As documented in the experiments, the error criterion is quite consistent to the basic characteristic of human visual system. © 2007 Wiley Periodicals, Inc. Int J Imaging Syst Technol, 16, 121,127, 2006. [source] Progressive edge detection compression for fingerprint imagesINTERNATIONAL JOURNAL OF IMAGING SYSTEMS AND TECHNOLOGY, Issue 5 2002Awad K. Al-Asmari In this article, a progressive fingerprint image compression (for storage or transmission) that uses an edge detection scheme is developed. First, the image is decomposed into two components: the primary component, which contains the edges, and the secondary component, which contains the textures and the features. Then, a general grasp for the image is reconstructed in the first stage at bit rates of 0.0223 and 0.0245 bpp for the tested fingerprints images (samples 1 and 2), respectively. The quality of the reconstructed images is competitive with the 0.75-bpp target bit set by the FBI standard. Also, the compression ratio and the image quality of this algorithm are competitive with other methods reported in the literature. The compression ratio for our algorithm is about 45:1 (0.180 bpp). © 2003 Wiley Periodicals, Inc. Int J Imaging Syst Technol 12, 211,216, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ima.10025 [source] Enhanced video streaming network with hybrid P2P technologyBELL LABS TECHNICAL JOURNAL, Issue 3 2008Yugu Zeng Hybrid peer-to-peer (P2P) approaches combine the cost advantages of P2P solutions with the quality and capacity characteristics of hosted content server solutions. Thousands of users have home devices providing many streaming and storage resources that promise superior scalability and adaptability over server-based content distribution networks. On the other hand, the involvement of these peers is encumbered by additional latency and resource management overhead. In this paper, we describe how the capacity advantage of hybrid P2P solutions over pure P2P solutions translates into higher achievable bit rates. The modeling and analysis of a heterogeneous architecture presented provides insight into the achievable system latency and design directions for topology resource management. The results indicate the potential of hybrid P2P approaches to serve as a cost-effective streaming platform with bit rates and latency quality close to those of other digital video distribution systems such as digital video broadcast- satellite (DVB-S) or Internet Protocol television (IPTV), but with better deployment flexibility for the future digital media "prosumer" models. © 2008 Alcatel-Lucent. [source] | ||||||||||