IEE Proceedings - Communications
Volume 152, Issue 3, June 2005
Volumes & issues:
Volume 152, Issue 3
June 2005
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- Author(s): N. Zhang ; Q. Shi ; A. Nenadic ; M. Merabti ; R. Askwith
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 257 –261
- DOI: 10.1049/ip-com:20045166
- Type: Article
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A protocol is presented for the fair digital-signature exchange commonly seen in e-commerce systems. The protocol incorporates a novel approach for the recovery of a signature with the assistance of an offline trusted third party that can impose a penalty on a misbehaving party. This new approach enables the protocol to achieve fairness in a simpler and more efficient manner than other relevant fair exchange protocols, while being able to deter a dishonest party from misbehaving. - Author(s): N.-Y. Lee and M.-F. Lee
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 262 –264
- DOI: 10.1049/ip-com:20045053
- Type: Article
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Kim, Shin and Kim have proposed an efficient metering scheme (2001) to measure the visits of clients to a web server. It has been found that the Kim–Shin–Kim metering scheme can suffer from a malicious forgery attack. An improved scheme is proposed to repair the security flaw. - Author(s): P. Azmi and F. Marvasti
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 265 –271
- DOI: 10.1049/ip-com:20045160
- Type: Article
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A novel decoding technique for DFT-based error control codes that is robust against quantisation noise and additive white Gaussian noise is proposed. The algorithm simultaneously determines the number and the positions of the corrupted samples. In contrast to the conventional decoding technique, the proposed decoding method is stable in the presence of both quantisation noise and additive white Gaussian noise. - Author(s): H. Wang ; S.N. Koh ; W.-W. Chang
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 272 –276
- DOI: 10.1049/ip-com:20040959
- Type: Article
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The performance of speech coders, such as the ITU-T G.722.1 wideband speech coder, that employ non-self-synchronising variable-length codes is greatly degraded when the received bit stream is in error. The authors study the use of reversible variable-length codes (RVLCs) to replace the Huffman codes recommended in the G.722.1 coder. A modified trellis-based scheme to decode the RVLCs-encoded bit stream is proposed and applied to the coder to improve its performance when bit errors occur. Simulation results show that the modified coder is much more robust in segmental SNR performance as compared to the original G.722.1 coder when bit errors occur. - Author(s): W. Hoteit ; Y.R. Shayan ; A.K. Elhakeem
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 277 –281
- DOI: 10.1049/ip-com:20045101
- Type: Article
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The effect of imperfect channel estimation on the bit error rate (BER) of multiple-input multiple-output communication systems utilising space–time coding is investigated. A multipath channel is considered and a given level for the channel estimation error is assumed. The receiver employs an equaliser to reduce the ISI in the received signal. A closed-form expression for the SNR at the output of the equaliser that employs the results of the channel estimation is derived. The theoretical SNR derived is used as a basis to assess the performance of the system. Results are applicable to any channel estimation technique. Analysis shows that the deterioration of performance in the multiple transmit antenna scheme outweighs the benefits achieved over the single antenna case when the SNR and channel estimation error are large. The degradation in the transmit diversity scheme exceeds 8 dB to achieve a BER of 10−4 when the channel estimation error is 5% relative to the perfect channel estimation case. - Author(s): A. Miri ; E. Hons ; A.K. Khandani
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 282 –287
- DOI: 10.1049/ip-com:20041038
- Type: Article
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The problem of optimising the structure of the encoder/decoder pair in a discrete communication system, with an additive distortion measure, is formulated in terms of a quadratic programming (QP) problem. This new formulation benefits from the following special features: it optimises the joint effects of the source/channel coding on the end-to-end distortion; and the encoder and the decoder structures are not restricted to being the inverse of each other. A method which obtains an ɛ-minimiser approximation of an optimum point of a general QP problem is discussed. Some simulation results based on this method are also given. - Author(s): H.C. Park
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 288 –294
- DOI: 10.1049/ip-com:20045036
- Type: Article
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A constant-envelope BPSK signal is realised by specifying the phase transition characteristics of a conventional BPSK signal. The CE-BPSK signal achieves greater power and bandwidth efficiency than a conventional square-root raised-cosine (SRRC) filtered BPSK signal, while still having backward compatibility with the conventional BPSK signal. Utilising the constant-envelope and continuous phase characteristics, it is shown that the CE-BPSK is a kind of continuous phase modulation. This allows the CE-BPSK signal to be generated in existing FM-based transmitter systems. The power spectral density and eye-diagram of the continuous phase modulated CE-BPSK signal are identical to those of the I/Q modulated CE-BPSK signal. It is shown that the bandwidth efficiency of the CE-BPSK signal is ten times greater than the SRRC-filtered BPSK signal in a switching amplification environment. It is also shown that the bit error rate performance of the CE-BPSK signal is better than that of the SRRC-filtered BPSK signal by 0.7 dB at a BER of 10−4 in the switching amplification environment. - Author(s): Y. Luo and S. Lambotharan
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 295 –301
- DOI: 10.1049/ip-com:20045048
- Type: Article
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A tap-constrained constant modulus algorithm is proposed for the blind equalisation of a time-reversal-based space–time block-coded (STBC) transmission. The tap constraint has been developed by exploiting certain algebraic properties of STBC, and this constraint avoids the problem of repeated retrieval of identical sources found in the conventional constant modulus algorithms. The bit-error-rate performance of the proposed blind detection algorithm is shown to be very close to that of a pilot-based Viterbi equaliser. - Author(s): H. Zhou ; W.L. Woo ; B.S. Sharif
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 302 –310
- DOI: 10.1049/ip-com:20045111
- Type: Article
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A new blind adaptive multiuser detection scheme based on a hybrid of Kalman filter and subspace estimation is proposed. It is shown that the detector can be expressed as an anchored signal in the signal subspace and the coefficients can be estimated by the Kalman filter using only the signature waveform and the timing of the desired user. The resulting subspace-based algorithm brings the benefit of lower computational complexity than the full-rank approach, and the theoretical analysis indicates that the proposed algorithm is also superior in convergence performance. The adaptive implementation in a dynamic environment such as a variable number of users is obtained by seamlessly integrating a subspace tracking algorithm. The new subspace-based method is effective in AWGN channels as well as in slowly time-varying Rayleigh fading channels. Moreover, the proposed detector is much more robust against the signalling waveform mismatch and inaccurate knowledge of the amplitude of the desired signal than the full-rank one, as demonstrated by computer simulations. - Author(s): Z. Li ; A. Das ; A.K. Gupta ; S. Nandi
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 311 –319
- DOI: 10.1049/ip-com:20041022
- Type: Article
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Emerging IEEE 802.11 standards provide very high raw bandwidth. However, the overheads introduced by the physical and MAC layers are also increasingly substantial. Moreover, the multi-rate capability provided by the various physical layers requires that the MAC layer adapts the transmission rate according to the channel conditions. Recently, several rate-adaptation algorithms have been proposed. However, all the schemes have considered the rate adaptation for the Data frame only, while assuming that the control frames are always transmitted at a low basic rate. Also, the rate adaptation function is performed either at the sender side or at the receiver side. A new rate-adaptation algorithm called the full auto rate (FAR) algorithm is proposed, which is able to adapt the transmission rate of all the frames (both control and Data). The FAR algorithm combines the sender- and receiver-based methods proposed in the literature, that is, the rate adaptation of the RTS/CTS frames is done at the sending side of these frames while that for the Data/ACK frames is done at the receiving side of the frames. Moreover, in order to cope with the issues involved in virtual carrier sensing (VCS), the authors have proposed a modified virtual carrier sensing (MVCS) mechanism. Both analytical and simulation results show that the FAR algorithm greatly improves the performance of IEEE 802.11. - Author(s): Y.-S. Yen ; W. Chen ; J.-C. Zhhuang ; H.-C. Chao
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 320 –326
- DOI: 10.1049/ip-com:20045139
- Type: Article
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p.
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Weighted fair queueing (WFQ) is a popular scheme because of its guaranteed bandwidth and bounded delay. To make WFQ more flexible, sliding weighted fair queueing (SWFQ) is proposed by combining priority-driven and share-driven scheduling for a real-time network. In this proposal, SWFQ can balance the share and priority-driven characteristics and can allow the application of WFQ to various network environments. This queueing algorithm can also be applied to either IPv4 or IPv6 networks as long as it can provide an adequate quality of service (QoS) support. - Author(s): T.T. Tay and Y. Zhang
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 327 –334
- DOI: 10.1049/ip-com:20041217
- Type: Article
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A peer-distributed web caching scheme with incremental update is proposed that allows a client to retrieve a ‘patch’ from the original server and the ‘patchable’ stale cache from a local or a peer cache storage. The stale cache together with the patch then regenerates the up-to-date file for the client. The protocol of the scheme is described and the benefits are analysed. Implementation issues are discussed. The experimental results show that most updates are minor and patches are generally smaller than the original file. The proposed scheme is efficient in reducing intercluster traffic and improving response time. - Author(s): M. Masugi ; T. Takuma ; M. Matsuda
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 335 –341
- DOI: 10.1049/ip-com:20045082
- Type: Article
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A method of assessing the quality of service (QoS) of TCP/IP-based video streams for real environments is proposed. To investigate the cause of video quality degradation and to observe the video QoS level at user clients, a method for monitoring the QoS-related parameters in both the transport and application layers for the target system is employed. To detect the cause of QoS degradation, the authors focus on the number of packet retransmissions and the transmission delay of packets observed in the transport layer. Also, for measuring the QoS level, the bit rate of video streams and buffer usage rates in the application layer are the focus. Based on measurement using an MPEG-4 video system in a real environment, it is confirmed that the buffer usage rate in the application layer tended to decrease before the detection of sudden decreases in the bit rate of video streams, and that underflow events where the buffer usage rate dropped to less than 30–50% tended to cause QoS deterioration of video applications. The proposed method deals with QoS-related parameters for TCP/IP-based video streams so it is possible to efficiently check the reasons for and the patterns of the QoS degradation. - Author(s): I.-S. Hwang ; W.-D. Tseng ; I.-F. Huang
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 343 –348
- DOI: 10.1049/ip-com:20045285
- Type: Article
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The 2×2 electro-optical directional coupler is a common switching element in dilated optical multistage interconnection networks (DOMINs) and can be applied to photonic switching technology. Because of the undesirable coupling within an optical switch, a portion of the energy of the input signal might be transmitted to the undesired output terminal in the switch. A switch is said to be faulty if it causes excessive crosstalk above a given threshold, and optical crosstalk is a problem commonly found in DOMINs. The problem of connections-scheduling with faulty switches in a dilated re-arrangeable photonic network, say dilated Benes network (DBN), is studied and an integrated fault tolerant DBN architecture is proposed. Based on the faulty switch information given from the fault diagnosis algorithm, the fault-free path generation algorithm and scheduling algorithm are devised to schedule a set of connections avoiding faulty switches with a polynomial time complexity in the DBN. It is desirable to keep the number of rounds as small as possible. The simulation results show that the proposed method outperforms the greedy-scheduling algorithm in terms of routing number, number of failed connections, minimum number of rounds, and connection success probability when the size of the network is large. Furthermore, the proposed methodology can be embedded in the optical switching system to achieve self-testing. - Author(s): L. Zhu ; N. Ansari ; J. Liu
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 349 –352
- DOI: 10.1049/ip-com:20045057
- Type: Article
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High-speed TCP (HSTCP) has been proposed to achieve high throughput in the high bandwidth and long-delay environment. The throughput of HSTCP is studied when the backbone network adopts the optical burst switching (OBS) technology. The study shows that small burst assembly time and a large burst loss ratio in OSB can reduce the throughput of HSTCP significantly, but HSTCP still performs much better than TCP. - Author(s): S.S. Chakraborty ; M. Liinaharja ; K. Ruttik
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 353 –356
- DOI: 10.1049/ip-com:20040549
- Type: Article
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Antenna diversity is a widely studied strategy to combat fading processes in wireless systems. While packet combining may further enhance receiver diversity, this issue has not received much attention. The authors present a study of ARQ schemes with different receiver diversity schemes. An important outcome of the work is that significant performance gain can be achieved with packet combining along with selection diversity. - Author(s): O. Väänänen ; J. Vankka ; K. Halonen
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 357 –362
- DOI: 10.1049/ip-com:20059014
- Type: Article
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In conventional GSM base station solutions, transmitted carriers are combined after power amplifiers. A GSM signal is a constant envelope signal which means that a power efficient nonlinear power amplifier can be used. However, in third generation systems the signal is no longer a constant envelope signal, which means that linearity requirements for the PA are increased. Also, in the case of multicarrier GSM or EDGE transmission the peak to average power of the signal becomes high if the carriers are combined in the digital domain. High linearity requirements lead to low power efficiency and therefore to high power consumption. In order to achieve good efficiency in the PA the PAR must be reduced, i.e. the signal must be clipped. In the paper the effect of a peak windowing clipping algorithm is studied in the cases of WCDMA, GSM and EDGE transmission. Also, an efficient way to implement the peak windowing algorithm is presented. - Author(s): K.K. Leung ; H.Y. Kwan ; C.W. Sung ; T.M. Lok
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 363 –370
- DOI: 10.1049/ip-com:20045017
- Type: Article
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The sequence adaptation problem for code division multiple access (CDMA) systems with antenna arrays is considered. The problem can be formulated as a sequence adaptation game. Each user would like to choose a signature sequence that minimises his/her interference based on an interference measure. It is well known that, with arbitrary interference measures, sequence adaptation in antenna array systems may not have an equilibrium point (Nash equilibrium). The authors prove that additivity, continuity and symmetry of the interference measures are three sufficient conditions for this existence. Adaptive algorithms are also developed for sequence adaptation in antenna array systems. Simulations show that sequence adaptation with antenna arrays can significantly improve the system throughput over the existing system with antenna arrays. - Author(s): A. Wagstaff and N. Merricks
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 371 –377
- DOI: 10.1049/ip-com:20045025
- Type: Article
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Currently, the levels of VHF and UHF man-made noise (MMN) in the electromagnetic environment are documented in the ITU-R Recommendation (2001), based on research undertaken in the 1970 s. In an effort to update this information, Mass Consultants Ltd. (MASS) conducted a measurement study on behalf of the Radiocommunications Agency (now Ofcom). The programme of work included the building of a Noise Measurement Facility (NMF) to collect and analyse MMN data between 100 MHz and 3 GHz with bandwidths up to 10 MHz. Analysis employed the Rayleigh and Weibull distributions to model white Gaussian noise (WGN) and impulsive noise (IN) components, respectively, through the generation of the amplitude probability distribution (APD). The results of this analysis suggest that the majority of MMN is comprised of WGN and that this level is higher than previous research would suggest. The IN component has also been modeled in a form that would be suitable for inclusion in the ITU-R standard on radio noise using just the mean and standard deviation of the Weibull distribution. - Author(s): R.C. Nongipur ; D.J. Shpak ; A. Antoniou
- Source: IEE Proceedings - Communications, Volume 152, Issue 3, p. 378 –384
- DOI: 10.1049/ip-com:20041211
- Type: Article
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A near-end crosstalk (NEXT) cancellation system that uses frequency-domain least-mean-squares (FDLMS) adaptive filters is proposed. The new system is equipped with a mechanism whereby the NEXT signals are detected and then FDLMS adaptive filters are assigned to cancel only the significant NEXT signals. The paper also explores various schemes of assigning step sizes to the adaptive filters. It has been found that by making the step sizes proportional to the magnitudes of the NEXT signals during the initial phases of adaptation, and equal later on, significant improvement in the convergence rate can be achieved. By returning after convergence to step sizes that are proportional to the magnitudes of the NEXT signals, a much better tracking performance can also be achieved.
Efficient fair digital-signature exchange based on misbehaviour penalisation
Secure and efficient web metering scheme
Robust decoding of DFT-based error-control codes for impulsive and additive white Gaussian noise channels
Application of reversible variable-length codes in robust speech coding
Effects of imperfect channel estimation on space–time coding performance
Optimising the combined source and channel coding of a discrete communication system
Power and bandwidth efficient constant-envelope BPSK signals and its continuous phase modulation interpretation
Tap-constrained constant modulus algorithm for the blind equalisation of time-reversal-based STBCs through the exploitation of algebraic properties
Subspace-based blind adaptive multiuser detection using Kalman filter
Full auto rate MAC protocol for wireless ad hoc networks
Sliding weighted fair queueing scheme for real-time applications
Peer-distributed web caching with incremental update scheme
QoS assessment of video streams over IP networks based on monitoring transport and application layer processes at user clients
Integrated fault tolerant connections-scheduling for dilated Benes network
Throughput of high-speed TCP in optical burst switching networks
Diversity and packet combining in Rayleigh fading channels
Simple algorithm for peak windowing and its application in GSM, EDGE and WCDMA systems
Stability issues of sequence adaptation for CDMA systems with antenna arrays
Man-made noise measurement programme
NEXT cancellation system with improved convergence rate and tracking performance
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