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Online ISSN 1751-8636 Print ISSN 1751-8628

IET Communications

Volume 4, Issue 8, 21 May 2010

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Volume 4, Issue 8

21 May 2010

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    • Unified scaling factor approach for turbo decoding algorithms
      • Author(s): D.-W. Yue  and  H.H. Nguyen
      • Source: IET Communications, Volume 4, Issue 8, p. 905 –914
      • DOI:  10.1049/iet-com.2009.0125
      • Type: Article
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      • p. 905 –914 (10)
        Turbo (iterative) decoding can be implemented with a large family of soft-input soft-output (SISO) algorithms, including suboptimal but practical algorithms such as the soft-output Viterbi algorithm (SOVA) and max-log-MAP algorithm. The performance with these practical algorithms can be improved with simple scaling factor methods. However, their theoretical analysis was mainly based on the relaxed assumption proposed by Papke et al. that the extrinsic information is Gaussian distributed. This study proposes a novel scaling factor approach for reducing both the overestimation of reliability values and the correlation between the intrinsic and extrinsic information. Explicit formulae for computing the scaling factors are derived based on mathematical statistics. A key difference compared to the scaling factor method of Papke et al. is that the proposed scaling factors can be computed off-line. The numerical results show that when implemented in the decoding algorithm of block component codes, the proposed scaling factor approach improves the performance of turbo decoding over additive white Gaussian noise and Rayleigh fading channels. It is superior than the method by Papke et al. in both performance and complexity.
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      Improved channel access protocol for cooperative ad hoc networks
      • Author(s): A. Basyouni ; W. Hamouda ; A. Youssef
      • Source: IET Communications, Volume 4, Issue 8, p. 915 –923
      • DOI:  10.1049/iet-com.2009.0461
      • Type: Article
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      • p. 915 –923 (9)
        Efficient utilisation of bandwidth and high data rates have a great impact on the performance of ad hoc wireless networks. The use of cooperative diversity, where neighbouring stations may act as relay nodes to transfer the source data to the desired destination node through an independent relay channel, has shown to provide diversity gain and consequently improve the achievable bit rate. However, this is usually attained at the expense of loosing some wireless resources such as bandwidth. On the other hand, the use of directional antennas has shown to offer an effective way for efficient bandwidth utilisation. Here, the authors propose a new channel access protocol for transmission over cooperative ad hoc networks. The proposed protocol, which aims at minimising the number of blocked nodes and consequently improving the system throughput, employs directional antennas at both the source and relay stations. The network performance under the proposed settings is modelled using continuous Markov chains. The steady-state transmission blocking probability and the average network throughput are obtained by analysing the derived Markov model. The analytical results, which are validated through simulations, show the improvement in performance compared to the carrier sense multiple access with collision avoidance protocol that employs omnidirectional antennas.
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      Problem of localisation in networks of randomly deployed nodes: asymptotic and finite analysis, and thresholds
      • Author(s): F. Daneshgaran ; M. Laddomada ; M. Mondin
      • Source: IET Communications, Volume 4, Issue 8, p. 924 –936
      • DOI:  10.1049/iet-com.2009.0553
      • Type: Article
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      • p. 924 –936 (13)
        Consider a two-dimensional domain S ⊆ ℜ2 containing two sets of nodes from two statistically independent uniform Poisson point processes with constant densities pL and pNL. The first point process identifies the distribution of a set of nodes having information about their positions, hereafter denoted as L-nodes (localised-nodes), whereas the other is used to model the spatial distribution of nodes that need to localise themselves, hereafter denoted as NL-nodes (not localised-nodes). For simplicity, both kinds of nodes are equipped with the same kind of transceiver, and communicate over a channel affected by shadow fading. As a first goal, the authors derive the probability that a randomly chosen NL-node over S gets localised as a function of a variety of parameters. Then, the authors derive the probability that the whole network of NL-nodes over S gets localised. As with many other random graph properties, the localisation probability is a monotone graph property showing thresholds. In this work, the authors derive both finite (when the number of nodes in the bounded domain is finite and does not grow) and asymptotic thresholds for the localisation probability. In connection with the asymptotic thresholds, the authors show the presence of asymptotic thresholds on the network localisation probability in two different scenarios. The first refers to dense networks, which arise when the domain S is bounded and the densities of the two kinds of nodes tend to grow unboundedly. The second kind of thresholds manifest themselves when the considered domain increases but the number of nodes grow in such a way that the L-node density remains constant throughout the investigated domain. In this scenario, what matters is the minimum value of the maximum transmission range averaged over the fading process, denoted as dmax, above which the network of NL-nodes almost surely gets asymptotically localised.
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      Multi-path transmission control scheme combining bandwidth aggregation and packet scheduling for real-time streaming in multi-path environment
      • Author(s): M.-F. Tsai ; N. Chilamkurti ; J.H. Park ; C.-K. Shieh
      • Source: IET Communications, Volume 4, Issue 8, p. 937 –945
      • DOI:  10.1049/iet-com.2009.0661
      • Type: Article
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      • p. 937 –945 (9)
        These days, a wide variety of wireless interfaces are available to connect to Internet. When coverage area of these different technologies overlap, receiver equipped with multiple interfaces can use them simultaneously to improve the performance of its applications in terms of bandwidth rely on bandwidth aggregation. However, specific conditions such as bottleneck bandwidth and end-to-end delay need to be accounted for before using such techniques. If this problem of end-to-end delay and bottleneck bandwidth are not properly addressed, there may be many packets along multiple paths which can arrive late and can lead to a large number of out-of-order packets at the receiver, which can eventually cause serious degradation of video quality at the receiver. For this reason, in this study, the authors propose a multi-path transmission control scheme (MTCS) combining bandwidth aggregation and packet scheduling for real-time streaming in a multi-path environment. In a bandwidth aggregation scheme, the authors propose a mathematical model to find the transmission rate over each path in order to obtain the optimal total throughput. However, the end-to-end delay of each path is not the same. The out-of-order packets problem will become serious in a multi-path environment. Therefore the authors propose a packet scheduling scheme to arrange the transmission sequence in order to effectively minimise the impact of packet reordering at the receiver. Our proposed control scheme not only aggregates the available bandwidth of multiple paths, but also reduces the time of packet reordering at the receiver. Experimental results show with our proposed scheme, the authors not only obtain the optimal transmission throughput but also reduce packet reordering delays under varying drop and delay conditions caused by the underlying network.
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      Achievable rate region for broadcast-relay networks with two cooperative relays
      • Author(s): S. Salehkalaibar ; L. Ghabeli ; M.R. Aref
      • Source: IET Communications, Volume 4, Issue 8, p. 946 –955
      • DOI:  10.1049/iet-com.2009.0555
      • Type: Article
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      • p. 946 –955 (10)
        The authors consider the problem of broadcast-relay-networks with two cooperative relays. There are a transmitter, two relays and two receivers in the network. The message of the transmitter intended to the receivers has common and private parts. The relays fully cooperate with each other and with the transmitter to send the common part of the message, whereas the private parts of the message are sent through the direct links between the transmitter and receivers. The authors found an achievable rate region for this network by using the symmetric relaying strategy. In this strategy, each relay completely decodes the message of other relay. In the proof the authors took advantage of regular encoding/sliding window decoding at relays and simultaneous backward decoding analysis at receivers. Marton's broadcast code construction is used at the transmitter to split the rate between the users. Three special cases of achievable rate region are shown: (i) Kramer's achievable rate region for broadcast relay channel; (ii) Ghabeli's achievable rate for symmetric two-relay network; and (iii) Marton's achievable rate region for broadcast channel with common message. The additive white Gaussian noise model is also considered and the achievable rate region of Gaussian network is discussed.
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      Optimal power allocation and power control for VBLAST systems with M-ary modulations
      • Author(s): H.-Y. Chen ; C.-H. Chuang ; P.-C. Yeh ; S.A. Zummo
      • Source: IET Communications, Volume 4, Issue 8, p. 956 –966
      • DOI:  10.1049/iet-com.2009.0381
      • Type: Article
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      • p. 956 –966 (11)
        In this study, the authors propose the optimal power allocation scheme that minimises the symbol error probability (SER) of vertical Bell Laboratories layered space-time (VBLAST) systems using M-ary modulations. The essence of the power allocation is to intentionally cause different received power from each layer at the receiver. By judiciously allocating the transmitting power to the layers, the performance of the successive interference cancelation at the receiver can be significantly improved. The exact SER of the VBLAST systems with non-uniform power allocation is analysed for the Rayleigh fading case. The SER is minimised to determine the optimal power allocation pattern for the VBLAST systems with M-ary modulations. Simulations show that our SER analysis is accurate and the proposed power allocation scheme significantly improves the performance of the VBLAST systems in fast fading environments by 3.5–4 dB. The work is applied to the power control of VBLAST systems in slow fading environments. Significant SNR gains of 8.5–10 dB are observed in the numerical experiments. It is also observed that the proposed optimal power allocation scheme can effectively reduce the SER variation among the layers.
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      Selective relaying schemes for distributed space–time coded regenerative relay networks
      • Author(s): C. Zhang ; J. Zhang ; H. Yin ; G. Wei
      • Source: IET Communications, Volume 4, Issue 8, p. 967 –979
      • DOI:  10.1049/iet-com.2009.0438
      • Type: Article
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      • p. 967 –979 (13)
        In distributed space–time (DST)-coded regenerative relay networks, demodulation error produced by relays degrades the receiver performance significantly. To mitigate this disadvantage, two threshold-based selective relaying schemes are proposed, that is, centralised selecting scheme and distributed selecting scheme, where each relay forwards signals only if its received signal–noise ratio is larger than a threshold. Both proposed schemes can work well with arbitrary modulation constellation and any number of relays and no matter whether the source–destination channel is available or not. Simulation results show both proposed selective relaying schemes outperform conventional schemes significantly and the improvement increases as the scale of relay network grows. Centralised selecting has a slightly better performance than the distributed selecting. However, the latter has a far lower system cost. This contribution provides two useful relaying mechanisms to mitigate error propagation.
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      Network congestion estimation using packet time series analysis
      • Author(s): L. Cong ; G. Lu ; Y. Chen ; B. Deng ; X. Li
      • Source: IET Communications, Volume 4, Issue 8, p. 980 –989
      • DOI:  10.1049/iet-com.2009.0506
      • Type: Article
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      • p. 980 –989 (10)
        Previously, the network must be congested by probing flow before the congestion-related parameters (such as background flow and available bandwidth) can be estimated, which lead to the inaccuracy and inefficiency of today's Internet. In this work, the authors study how to estimate network state without saturating the network. By introducing the queueing theory, this study proposes a novel packet time series analysis (PTSA) framework model, which can be used to estimate the congestion-related parameters without saturating the network. The accuracy and efficiency of PTSA are validated under NS-2 simulation environment. The performance of PTSA methodology is evaluated in Schooner test-bed with a special scenario. The analytical, simulative and experimental results show that PTSA framework is more efficient to estimate network state with less aggression and higher sensitivity than those methods that need to saturate the network.
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      Blind carrier frequency offset estimator for multi-input multi-output-orthogonal frequency division multiplexing systems over frequency-selective fading channels
      • Author(s): S. Younis ; A. Al-Dweik ; A. Hazmi ; B. Sharif ; C. Tsimenidis
      • Source: IET Communications, Volume 4, Issue 8, p. 990 –999
      • DOI:  10.1049/iet-com.2009.0477
      • Type: Article
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      • p. 990 –999 (10)
        This study presents a new blind carrier frequency offset (CFO) estimation technique for multi-input multi-output (MIMO) orthogonal frequency division multiplexing (OFDM) systems employing space–time coding (STC). CFO estimation is crucial for OFDM systems to avoid the performance degradation because of the inter-carrier interference that results when the CFO is not estimated and compensated accurately. Based on the assumptions that the data symbols are selected from a constant modulus constellation and the channel is varying slowly over time, a new blind CFO estimator is proposed by minimising the power difference between all subcarriers in two consecutive STC blocks. Therefore the proposed system exploits all subcarriers in time and frequency domain, which provides a remarkable performance improvement over other techniques reported in the literature. The complexity of the proposed estimator is substantially reduced by approximating the cost function by a sinusoid that can be minimised using direct closed-form computations within one OFDM symbol period. Monte Carlo simulations are used to assess the performance of the proposed system by means of mean squared error (MSE) in both static and time-varying frequency-selective fading channels. The simulation results demonstrate that the proposed estimator can eliminate the MSE error floors that usually appear at moderate and high signal-to-noise ratios for the estimators that work only in frequency domain.
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      Low-complexity joint data detection and channel equalisation for highly mobile orthogonal frequency division multiplexing systems
      • Author(s): H. Doğan ; E. Panayırcı ; H. Vincent Poor
      • Source: IET Communications, Volume 4, Issue 8, p. 1000 –1011
      • DOI:  10.1049/iet-com.2009.0522
      • Type: Article
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      • p. 1000 –1011 (12)
        This study is concerned with the challenging and timely problem of channel equalisation and data detection for orthogonal frequency division multiplexing (OFDM) systems in the presence of frequency-selective and very rapidly time-varying channels. The algorithm is based on the space alternating generalised expectation-maximisation (SAGE) technique which is particularly well suited to multicarrier signal formats and can be easily extended to multi-input multi-output-OFDM systems. In fast fading channels, the orthogonality between subcarriers is destroyed by the time variation of a fading channel over an OFDM symbol duration which causes severe inter-carrier interference (ICI) and, in conventional frequency-domain approaches, results in an irreducible error floor. The proposed joint data detection and equalisation algorithm updates the data sequences in series leading to a receiver structure that also incorporates ICI cancellation, enabling the system to operate at high vehicle speeds. A computational complexity investigation as well as detailed computer simulations indicate that this algorithm has significant performance and complexity advantages over existing suboptimal detection and equalisation algorithms proposed earlier in the literature.
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      Cross-layer scheduling for multiuser H.264 video transmission over wireless networks
      • Author(s): F. Li ; G. Liu ; L. He
      • Source: IET Communications, Volume 4, Issue 8, p. 1012 –1025
      • DOI:  10.1049/iet-com.2009.0618
      • Type: Article
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      • p. 1012 –1025 (14)
        The problem of the scheduling of the multiuser H.264 video transmission over wireless networks is addressed in this work. A cross-layer algorithm, including the application (APP) layer, the media access control (MAC) layer and the physical (PHY) layer, is presented. More especially, a compressed method is proposed to estimate the video quality contribution caused by each packet in the APP. The objective is to maximising video qualities by satisfying the delay constraint and the fairness among all users. The algorithm is compatible with the emerging wireless standards, such as IEEE 802.16 and high speed downlink packet access (HSDPA). Simulations show that the proposed scheme significantly outperforms the conventional scheduling schemes in terms of received video qualities as well as the fairness properties.
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      Scalable voice over Internet protocol service-level agreement guarantees in converged transmission control protocol/Internet protocol networks
      • Author(s): Q. Yang  and  J.M. Pitts
      • Source: IET Communications, Volume 4, Issue 8, p. 1026 –1035
      • DOI:  10.1049/iet-com.2009.0491
      • Type: Article
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      • p. 1026 –1035 (10)
        Increasing complexity of the recent introduced quality of service (QoS) provisioning tools in packet-switched networks becomes a major obstacle for Internet service providers to practically deploy and manage these mechanisms to guarantee a range of transparent real-time service-level agreements. The authors revisit the existing Internet protocol (IP) QoS approaches and present a cost-effective solution to provide voice over Internet protocol (VoIP) service guarantees in converged IP networks by combining open shortest-path first traffic engineering with new insights for the configuration of standard active queue management random early detection mechanism. Extensive simulation studies are carried out, and the service quality is explicitly quantified using international telecommunication union E-model for a range of scenarios representing different types of network uncertainties. The direct representation of VoIP service quality well demonstrate that such a simple, coordinated approach, in keeping with the Internet paradigm, can achieve increased load for a given quality level and greater resilience under degraded network conditions.
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