IEE Proceedings - Communications
Volume 153, Issue 2, April 2006
Volumes & issues:
Volume 153, Issue 2
April 2006
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- Author(s): S. Lambotharan ; Y. Luo ; C. Toker ; S.R. Alty ; M. Gani
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 159 –164
- DOI: 10.1049/ip-com:20050320
- Type: Article
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A blind multi-user detection algorithm for space-time block coded (STBC) signals is proposed based upon subspace methods and constant modulus algorithms for multiple antenna based receivers. The subspace method is used to determine blindly the multiuser detector required for the mitigation of multi-user interference and to coherently combine multipath and multiple receive antenna signals. It is shown that this subspace method for STBC could however, solve this problem only up to a matrix ambiguity and the structure of this ambiguity matrix looks like a channel matrix obtained for a single user STBC system with frequency non-selective channel. Hence, the remaining problem is shown to be equivalent to blind equalisation of a single user STBC in flat fading channels. A tap constrained constant modulus algorithm is proposed to solve the remaining problem. In this blind method, only the signature waveform of the user of interest is assumed to be known, and the signature waveforms of all other users in addition to the channel impulse responses of all the users are assumed to be unknown. Simulation results confirm superior performance of the proposed method over conventional non-blind matched filters. - Author(s): S.-L. Ng
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 165 –168
- DOI: 10.1049/ip-com:20050073
- Type: Article
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The concept of quasi-threshold multipartite access structures for secret sharing schemes is introduced. These access structures allow one distinguished class of participants to retain control over the reconstruction of the secret, while allowing flexibility over the quorum of participants from other classes. While ideal bipartite access structures have been classified, the case for multipartite access structures is still open. Here it is shown that quasi-threshold multipartite access structures are ideal by constructing a projective space representation of the associated matroids. - Author(s): C.-Y. Ho and Y.-C. Chen
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 169 –176
- DOI: 10.1049/ip-com:20050184
- Type: Article
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Improving the performance of the traditional TCP in wireless IP communications has been an active research area. The significant cause of packet losses in such heterogenous networks is no longer limited to network congestion. The performance degradation of TCP in wireless and wired-wireless hybrid networks is mainly due to its lack of ability to differentiate the packet losses caused by network congestions from the losses caused by wireless link errors. New variants of TCP Vegas and TCP Reno named Snug-Vegas and Snug-Reno, respectively, are proposed. By using random-loss indications marked by base stations, Snug-Vegas and Snug-Reno may detect random packet losses precisely. Through the packet loss differentiation, Snug-Vegas and Snug-Reno react appropriately to the losses, and based on the simulation results, it is seen that the throughput of connection over heterogeneous networks can be significantly improved. - Author(s): J. Zeng ; L. Zakrevski ; N. Ansari
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 177 –182
- DOI: 10.1049/ip-com:20045286
- Type: Article
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The proportional differentiation service model has emerged as a refined version of the DiffServ qualify of service (QoS) architecture. It relies on a series of parameters to enforce proportionally differentiated QoS criteria, such as queueing delay and packet loss. From the perspective of proportional loss differentiation, a large amount of work has been done on carrying out loss differentiation based on given parameters. Under certain network conditions, however, the loss differentiation cannot be met based on these hand picked parameters. While existing work focuses on enhancing dropping mechanisms themselves to honour the loss differentiation, the paper looks into calculating feasible differentiation parameters. By forming an optimisation problem based on multiple class blocking thresholds, the paper introduces a simple quantitative guideline to compute loss differentiation parameters. Derived closely related to network statuses and packet dropping mechanisms, these parameters ease the difficulty that dropping mechanisms may encounter when enforcing packet loss differentiation. Its finite computation time, moreover, makes practical implementation possible. Analytical and numerical results are given to substantiate the new approach and its merits. - Author(s): F. Alharbi and N. Ansari
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 183 –188
- DOI: 10.1049/ip-com:20045232
- Type: Article
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The resilient packet ring (RPR), defined under IEEE 802.17, has been proposed as a high-speed backbone technology for metropolitan area networks. RPR is introduced to mitigate the underutilisation and unfairness problems associated with the current technologies SONET and Ethernet, respectively. The key performance objectives of RPR are to achieve high bandwidth utilisation, optimum spatial reuse on the dual rings, and fairness. The RPR standard implements three traffic classes: Class A, Class B, and Class C. The RPR MAC has one queue for each traffic class. A potential performance limitation is associated with the head-of-line blocking. When the MAC uses a single FIFO to buffer frames awaiting access, a packet that is traversing through a congestion point may block transmission of other packets destined to a point before the congestion. The use of virtual destination queues (VDQs) to avoid the head-of-line blocking is introduced. Different bandwidth allocation policies are discussed to assign rates to VDQs. Finally, a bandwidth allocation policy is proposed, which would achieve the maximum utilisation at a very low complexity. - Author(s): T. Kocak
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 189 –194
- DOI: 10.1049/ip-com:20050482
- Type: Article
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Performance evaluation methods that have been used so far failed to capture the real characteristics of network traffic. This is especially true when the service times are general such as the variable bit rate (VBR) video, which is proven to be self-similar. In the paper, diffusion approximation methods are proposed to analyse a dynamic scheduler for self-similar VBR video traffic in asynchronous transfer mode (ATM) networks. The proposed scheme employs a coupled queueing system that has been used extensively in the modelling of computer and communication systems. Diffusion approximations methods are used to decouple a queueing system which represents an ATM network node into separate G/G/1 queues. Real MPEG video traces are used in the discrete event simulation. Results are compared with the approximation, and are found to work very well under different traffic conditions. - Author(s): A.C.M. Fong and A. Simpson
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 195 –200
- DOI: 10.1049/ip-com:20050462
- Type: Article
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Variable length codes (VLC) have found widespread use for efficient encoding of symbols with unequal probabilities in many practical situations. However, bit errors can cause a loss of decoder synchronism that often leads to error propagation. Recent advances in the development of VLC that limit the effect of error propagation, such as reversible VLC and self-synchronising VLC, contribute further to their popularity. However, it is still important to model the performance of various VLC code sets so that comparisons can be made between different coding schemes and for different applications. Communicating sequential processes (CSP) is a well-established formal description technique that allows one to analyse the behaviour of concurrently evolving processes. It has been successfully applied to industrial-scale problems. The paper describes how CSP can be applied to the modelling of the resync process of VLC. This approach offers three main advantages. First, a complete model comprises processes and events as basic building blocks, making this approach highly scalable. Second, different semantic models provide different level of abstraction. Third, tools are available for automatic generation of sync sequences, which also provide a measure of sync performance. This facilitates comparisons of different VLC schemes in addition to code sets obtained using a given coding scheme. - Author(s): I. Elhanany and B. Matthews
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 201 –204
- DOI: 10.1049/ip-com:20050331
- Type: Article
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A novel performance analysis of output queued cell switches that are introduced with general independent heterogeneous traffic is presented. Random arbitration is employed whereby non-empty queues compete equally for service within each switching interval. In particular, the case of bursty two-state Markov-modulated arrivals is studied in which input ports generate bursty streams that are non-uniformly distributed. Under the assumption of a memoryless server, the probability generating function of the interarrival process is utilised to derive an approximation for the queue size distribution. The methodology established forms a flexible tool in obtaining bounds on the behaviour and expected performance characteristics of output queued switches under a wide range of correlated traffic scenarios. The validity of the analytical inference is established through simulation results. - Author(s): H. Mellah and F.M. Abbou
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 205 –206
- DOI: 10.1049/ip-com:20050550
- Type: Article
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A contention resolution scheme for slotted optical packet switching networks is proposed. The optical switch is augmented using auxiliary output ports connected to an auxiliary switch through fibre delay lines. The node architecture is illustrated and the results of the model show that packet blocking probability is improved using the proposed scheme. - Author(s): J. Xie ; A. Das ; S. Nandi ; A.K. Gupta
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 207 –212
- DOI: 10.1049/ip-com:20045271
- Type: Article
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207
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Broadcasting is one of the essential communication models of MANETs. Many MANET multicast routing protocols rely heavily upon MAC layer's broadcast service for data delivery, multicast architecture construction and maintenance. However, the broadcast mechanism of the IEEE 802.11 standards cannot provide reliable broadcasting support, and therefore, satisfactory performance of the multicast protocols cannot be guaranteed. An extension to IEEE 802.11 broadcast mechanism, called round-robin acknowledge and retransmit, to address this shortcoming is proposed. Different from other mechanisms, we employ a simple and effective acknowledge mechanism, in which the lost frames are reported by neighbouring nodes in a round-robin style to avoid the notorious ack explosion problem. A MAC layer retransmission scheme is then provided for the lost frames. By varying the traffic load and group size, under the on demand multicast routing protocol (ODMRP), extensive simulations show that the proposed scheme provides a highly reliable broadcast service to the routing layer, and ODMRP's performance is improved significantly. - Author(s): N.Y. Ermolova
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 213 –218
- DOI: 10.1049/ip-com:20045178
- Type: Article
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p.
213
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The clipping-filtering method is a simple and frequently used in practice technique for reducing envelope fluctuations in multicarrier systems. When analysing systems that use clipping-filtering, very often a linear amplification is assumed. Meanwhile, a nonlinear amplifier inserts additional distortions into the clipped-filtered multicarrier signal. The cascade of the clipper-filter-nonlinear amplifier results in a nonlinear system with memory. The author studies both time-domain and frequency domain effects in this system. It is shown that while aggravating the total degradation in the system (almost for all values of the clipping ratio), the procedure of clipping-filtering provides a reduction of the out-of-band power emission after a nonlinear amplification. The author studies the structure of the clipped-filtered-amplified signal and proves that the signal constellation after this procedure is similar to that obtained after memoryless nonlinear processing, i.e. the observed signal is an attenuated version of the useful signal corrupted by Gaussian-like noise. Moreover, even under severe clipping and the working points of the amplifier close to the saturation (i.e. small values of the amplifier back-offs), Bussgang's theory can be approximately used for the prediction of distortions in the above scheme. - Author(s): S. Yousefi
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 219 –232
- DOI: 10.1049/ip-com:20050008
- Type: Article
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The error probability of maximum-likelihood (ML) soft-decision decoded binary block codes rarely accepts exact closed forms. In addition, for long codes ML decoding becomes prohibitively complex. Nevertheless, bounds on the performance of ML decoded systems provide insight into the effect of system parameters on the overall system performance in addition to a measure of efficiency of the sub-optimum decoding methods used in practice. In the article, a comprehensive study of a number of lower and upper bounds on the error probability of ML decoding of binary codes over AWGN channel is provided. Bounds considered here are bounds based on the so-called Bonferroni-type inequalities and bounds developed primarily in the light of the geometrical structure of the underlying signal constellations. The interrelationships among the bounds are explored and current tightest bounds at different noise levels are pointed out. - Author(s): S.A. Zummo and W.E. Stark
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 233 –237
- DOI: 10.1049/ip-com:20045272
- Type: Article
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I-Q trellis codes are known to increase the time diversity of coded systems. When I-Q codes are used with multiple transmit antennas, the decoding and performance evaluation requires the construction of the high-complexity super-trellis of the component codes. In the paper, the bit error probability and the design criteria of I-Q ST codes are derived using the transfer functions of the component codes. Conditions for the geometrical uniformity of I-Q space-time (ST) codes are derived from the geometrical uniformity of the component codes. In addition, a low-complexity iterative receiver for I-Q ST codes is presented. The receiver essentially performs iterative detection and decoding. Results show that three iterations of the iterative receiver performs very close to the optimal decoding. - Author(s): H. Hagirahim
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 238 –244
- DOI: 10.1049/ip-com:20050062
- Type: Article
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By means of an overload analysis and obtaining an approximated expression for the duration of overload, it is shown that the probability of packet loss owing to buffer overflow for a packetised voice system with silence suppression can be approximated by a simple mathematical expression. - Author(s): A. Freedman ; Y. Rahamim ; A. Reichman
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 245 –255
- DOI: 10.1049/ip-com:20050066
- Type: Article
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The use of short bursts transmissions in modern communication systems is increasing as the use of packet data communication replaces traditional, continuous bit stream data communication. Turbo codes, which revolutionised communication coding, enable the operation of receivers in low signal-to-noise ratio (SNR) conditions, thus increasing the range of operation of the communication system. However, carrier frequency and phase synchronisation, needed for optimal coherent performance of the receiver, still remains an open problem at low SNR conditions for short bursts. A new efficient carrier synchronisation method for turbo coded short packet communication operating at low SNR values is presented. This method is based on maximising a newly invented objective function, the MSSO(Δf, φ) function, which uses the turbo decoder's soft decision outputs to iteratively improve the carrier synchronisation. This method is suitable for non-data-aided acquisition and tracking of the carrier frequency and phase offsets. It is shown, via simulations, that this algorithm achieves the lower bound on bit-error-rate performance of the turbo decoder at very low SNR values. - Author(s): R. Horan ; C. Tjhai ; M. Tomlinson ; M. Ambroze ; M. Ahmed
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 256 –262
- DOI: 10.1049/ip-com:20050415
- Type: Article
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It is shown how to construct an algorithm to search for binary idempotents that may be used to construct binary LDPC codes. The algorithm, which allows control of the key properties of sparseness, code rate and minimum distance, is constructed in the Mattson-Solomon domain. Examples are given of the codes constructed that include equivalent codes to the Euclidean and Projective Geometry codes in addition to some new codes. Codes having cycles of length 4 can also be constructed and are demonstrated to have good performance under iterative decoding. - Author(s): T. Ratnarajah
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 263 –271
- DOI: 10.1049/ip-com:20045192
- Type: Article
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The capacity of multiple-input multiple-output (MIMO) wireless communication systems over spatially correlated Rayleigh distributed flat fading channels with complex Gaussian additive noise is investigated. Specifically, the probability density function of the mutual information between transmitted and received complex signals of MIMO systems is derived. Using this density the closed-form ergodic capacity (mean), delay-limited capacity, capacity variance and outage capacity formulas for spatially correlated channels are derived and then these formulas are evaluated numerically. Numerical results show how the channel correlation degrades the capacity of MIMO communication systems. It is also shown that the density of mutual information of correlated/uncorrelated MIMO systems can be approximated by a Gaussian density with derived mean and variance, even for a finite number of inputs and outputs. - Author(s): S. Antoniou ; L. Christofi ; P.R. Green ; G.F. Gott
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 272 –278
- DOI: 10.1049/ip-com:20050124
- Type: Article
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The paper investigates the use of differential modulation techniques to achieve high rate data transmission by near vertical incidence sky wave high frequency propagation, using a 2.7 kHz voice channel bandwidth. Channel characteristics have been measured for a range of frequencies between 2.8 and 9.4 MHz, using a transmitter power of 100 W over a distance of 160 km. These measurements are used to characterise simulated channel conditions, and the simulator is then used to determine the performances of differential modulation methods. In particular, time-differential amplitude and phase shift keying and frequency-differential amplitude and phase shift keying modulation methods are investigated. Symbol error rates are presented for data rates ranging from 4.3 to 20.6 kbit/s, where the system performances are limited by time-varying channel dispersion rather than by additive noise. - Author(s): Y.C. Jung and J.W. Atwood
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 279 –287
- DOI: 10.1049/ip-com:20050082
- Type: Article
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Two algorithms have been used as the basis for playout of packet-based voice. The first one identifies jitter level using the metric of the difference between the observed network delay for a new packet and the estimated average for previous packets. The new value of the estimated jitter level is a weighted combination of the previous value of the estimated jitter level and the new value of the observed jitter level. The second one is based on the difference in the arrival times of successive packets. The value of α used in the weighted combination is normally fixed. ThE paper proposes the use of two values of the weighting factor α to make the estimate of jitter level react quickly when jitter increases, but react slowly when it decreases. Four playout algorithms are evaluated using several jitter patterns obtained experimentally for 30 paths. The results indicate that the dynamic playout algorithm using αLOW=0.995 and αHIGH=0.998002 can produce significant reductions in the lateness loss rates, especially for high jitter conditions. It is also shown that the interarrival time jitter metric used in the dynamic playout algorithm results in better lateness loss performances especially in low jitter conditions. Finally, given the tendency to lower network delay, it is shown that increasing the buffering delay permits achieving significantly lower lateness loss, while still maintaining acceptable overall delay. - Author(s): M. Ferrari ; S. Bellini ; A. Spalvieri
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 288 –294
- DOI: 10.1049/ip-com:20050292
- Type: Article
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The paper deals with coding for the band-limited AWGN channel. Coded modulation schemes based on high rate unpunctured turbo codes are proposed. It is shown that, owing to efficient decoding based on the trellis of the dual code and to the optimised code design, coded modulations based on this class of turbo codes allow excellent performance with moderate complexity for practical values of block size and redundancy. A mapping rule already envisaged in previously published papers, based on the partition ℤ/2lℤ with Gray mapping of l bits is exploited. Schemes based on the partition chain ℤ/2ℤ/4ℤ are also presented in a comparative way. From this comparison it is concluded that, when constraints are imposed on the number of dimensions and on the redundancy of the coded modulation, partial Gray mapping based on ℤ/4ℤ should be preferred to pure mapping by set partitioning and, with QAM constellations with more than 16 points, it should also be preferred to pure Gray mapping. - Author(s): J.A. Hernández and I.W. Phillips
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 295 –304
- DOI: 10.1049/ip-com:20050335
- Type: Article
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Traces collected at monitored points around the Internet contain representative performance information about the paths their probes traverse. If processed appropriately, basic measurement attributes, such as delay and loss, can be used to output conclusions about the performance status of the network, with subsequent applications in fault and performance management, network provisioning, traffic engineering and performance prediction. However, the task of analysis and extracting such valuable information from measurements only remains challenging. The Weibull mixture model, a method to characterise end-to-end network delay measurements within a few simple, accurate, representative and handleable parameters using a finite combination of Weibull distributions is presented. The model parameters are related to meaningful delay characteristics, such as average peak and tail behaviour in a daily profile, and can be optimally found using an iterative algorithm known as expectation maximisation. Studies on such parameter evolution can reflect current workload status and all possible network events impacting packet dynamics, with further applications in network management. The model is further tested and validated with real GPS synchronised measurements taken across the Internet, donated by RIPE NCC. - Author(s): W. Jia and J. Wang
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 305 –312
- DOI: 10.1049/ip-com:20045176
- Type: Article
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Sensors typically have limited power of transmission. Thus desired connectivity probability (CP) between the sensors is critical in the given deployment area. The paper presents a novel approach for analysing the lower bound of CP for sensor networks under uniformly distribution of the sensors in a given area. Initially, the area is divided into a grid (mesh) of blocks. The CP for each small block is then calculated and the small blocks are aggregated into a large one and the desired CP for the entire network is derived progressively. More specifically, given n sensors in each block, the CP is derived by (1) computing the CP of two adjacent blocks using a geometrical probabilistic approach when n=1, which is the precise result; (2) based on Equation 1, the CP for n>1 is derived, which is very close to the simulation results within an error of 1%; (3) progressively deriving the CP of an entire network through aggregations of the small blocks. Simulation results demonstrate the feasibility of the algorithm. - Author(s): A. Payandeh ; M. Ahmadian ; M. Reza Aref
- Source: IEE Proceedings - Communications, Volume 153, Issue 2, p. 313 –316
- DOI: 10.1049/ip-com:20050086
- Type: Article
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Both security and error control coding are very extensive subjects, each with a variety of sub-disciplines. A secure channel coding (joint encryption-channel coding) scheme provides both data secrecy and data reliability in one process to combat problems in an insecure and unreliable channel. In this paper, an adaptive secure channel coding scheme based on serial or parallel concatenated turbo codes is developed. Recent results indicate that the turbo principle delivers near-to-optimal strategies for the channel coding. Reliability and security are achieved by adapting the pseudo-random puncturing strategy to the conditions of the noisy channel. Simulation results show the relevance and superior performance of the proposed scheme at all signal-to-noise ratio levels.
Blind multiuser detection of STBC signals using hybrid constant modulus and subspace methods
Ideal secret sharing schemes with multipartite access structures
Snug-Vegas and Snug-Reno: efficient mechanisms for performance improvement of TCP over heterogeneous networks
Computing the loss differentiation parameters of the proportional differentiation service model
SSA: simple scheduling algorithm for resilient packet ring networks
Approximate analysis of a dynamic scheduler for self-similar video traffic in ATM networks
Using CSP to model the synchronisation process of variable length codes
On the performance of output queued cell switches with non-uniformly distributed bursty arrivals
Contention resolution in slotted WDM optical packet switching networks using cascaded auxiliary switch
Improving the reliability of IEEE 802.11 broadcast scheme for multicasting in mobile ad hoc networks
Nonlinear amplifier effects on clipped-filtered multicarrier signals
Performance evaluation of maximum-likelihood decoded linear binary block codes in AWGN interference using Bonferroni-type and geometrical bounds
Performance analysis and iterative decoding of I-Q trellis space-time codes
Expression for the probability of packet loss owing to buffer overflow for multiplexing packetised voice
Maximum-mean-square soft-output (M2S2O): a method for carrier synchronisation of short burst turbo coded signals
Idempotents, Mattson-Solomon polynomials and binary LDPC codes
Spatially correlated multiple-antenna channel capacity distributions
High rate data transmission in the mid-latitude NVIS HF channel
Dynamic adaptive playout algorithm using interarrival jitter and dual use of α
Coded modulation schemes based on partial Gray mapping and unpunctured high rate turbo codes
Weibull mixture model to characterise end-to-end Internet delay at coarse time-scales
Analysis of connectivity for sensor networks using geometrical probability
Adaptive secure channel coding based on punctured turbo codes
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