IEE Proceedings - Communications
Volume 151, Issue 3, June 2004
Volumes & issues:
Volume 151, Issue 3
June 2004
-
- Author(s): K. Warwick ; M. Gasson ; B. Hutt ; I. Goodhew ; P. Kyberd ; H. Schulzrinne ; X. Wu
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 185 –189
- DOI: 10.1049/ip-com:20040409
- Type: Article
- + Show details - Hide details
-
p.
185
–189
(5)
A signalling procedure is described involving a connection, via the Internet, between the nervous system of an able-bodied individual and a robotic prosthesis, and between the nervous systems of two able-bodied human subjects. Neural implant technology is used to directly interface each nervous system with a computer. Neural motor unit and sensory receptor recordings are processed real-time and used as the communication basis. This is seen as a first step towards thought communication, in which the neural implants would be positioned in the central nervous systems of two individuals. - Author(s): D. Pao
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 190 –196
- DOI: 10.1049/ip-com:20040322
- Type: Article
- + Show details - Hide details
-
p.
190
–196
(7)
The performance of the TCP over ATM networks is investigated. A TCP packet is segmented into multiple ATM cells at the sending node and reassembled at the receiving node. When congestion occurs, the ATM switch may drop some of the cells. If one or more cells of a TCP packet are dropped, the TCP packet is said to be corrupted. A corrupted packet will be discarded at the receiving node. Transmitting the cells of a corrupted packet will only waste network resources and lower the effective throughput. A major objective of a selective packet discard scheme is to minimise the packet fragmentation problem. Previous approaches such as the early packet discard (EPD) scheme and its variants may drop incoming cells when congestion is anticipated. As a result, the buffer is under-utilised most of the time. Packets may be dropped unnecessarily, and the fragmentation problem cannot be avoided. A new packet discard scheme, called the on-demand packet discard (ODPD) scheme, is presented by the author. The ODPD scheme can fully utilise the available buffer space and effectively avoid the fragmentation problem. The performance of ODPD is found to be better than EPD and its variants under all traffic conditions. - Author(s): G. Cerri ; R. De Leo ; D. Micheli ; P. Russo
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 197 –203
- DOI: 10.1049/ip-com:20040146
- Type: Article
- + Show details - Hide details
-
p.
197
–203
(7)
The authors present a method for planning a base station's position in a mobile communication system taking into account both the requirement to minimise the environmental impact of the radiated electromagnetic fields and the requirement to assure a suitable quality of service, i.e. C/I ratio, coverage, efficiency, served traffic. The model is based on five functionals and the overall optimisation procedure is carried out by a genetic algorithm. As an example of its application, the proposed method is applied to an imaginary town, subdivided into areas with different constraints for the previously mentioned requirements. Results reported show the behaviour of each functional, as well as the global optimisation of the network. - Author(s): M. Asvial ; R. Tafazolli ; B.G. Evans
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 204 –209
- DOI: 10.1049/ip-com:20040291
- Type: Article
- + Show details - Hide details
-
p.
204
–209
(6)
Novel strategies for automatic satellite constellation design with satellite diversity and radio resource management are proposed. The automatic satellite constellation design means that some parameters of satellite constellation design can be determined simultaneously. The total number of satellites, the altitude of a satellite, the angle between planes, the angle shift between satellites and the inclination angle are considered in the design. Satellite constellation design is modelled using a multiobjective genetic algorithm. This method is applied to low Earth orbit (LEO), medium Earth orbit (MEO) and hybrid constellations. The use of a genetic algorithm allows automatic satellite constellation design while achieving dual satellite diversity statistics. Furthermore, a strategy for dynamic channel allocation is proposed that uses a genetic algorithm for use in mobile satellite systems (MSS) networks. The main idea behind this algorithm is to use the minimum cost as a metric to provide optimum channel solutions for specified interference constraints. The simulation is designed for a MEO satellite constellation. Using this algorithm, the proposed model outperforms conventional dynamic channel assignment (DCA) schemes in terms of call blocking and call dropping probability. Generally, genetic algorithms are robust to dynamic variations in satellite constellation design and provide resource allocation improvements in DCA in MSS networks. - Author(s): W. Shi and M.D. Macleod
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 210 –216
- DOI: 10.1049/ip-com:20040145
- Type: Article
- + Show details - Hide details
-
p.
210
–216
(7)
The data rate of a digital communication system depends on the bandwidth available to the system. Nyquist's theorem states that it is possible to transmit without intersymbol interference (ISI) at 2B symbols/s, given a bandwidth of B Hz. At present, no commercially available system achieves this Nyquist efficiency. It is well known that under certain orthogonality conditions, data transmission using multiple, spectrally overlapping signals can achieve the Nyquist efficiency. In practice, orthogonality is difficult because distortion in a physical channel, generally unknown at the receiver, spoils orthogonality; and even for an ideal, distortionless channel, orthogonality requires strict synchronisation of symbol time, frequency and phase in the system. The performance of a spectral overlap system in point-to-multipoint (PMP) networks is investigated. The orthogonality conditions are partly satisfied to minimise interferences, but conditions that are difficult to achieve are relaxed: therefore interference is present. Joint receivers are proposed to combat this interference. The optimum maximum likelihood sequence estimation (MLSE) receiver is derived, and it is shown that orthogonal system performance can be achieved. Also, simpler, sub-optimum receivers are proposed, and it is shown by adjusting system parameters that they too can achieve near orthogonal performance at a fraction of the cost of the MLSE receiver. - Author(s): K. Choi ; Y. Chae ; J. Park
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 217 –220
- DOI: 10.1049/ip-com:20040321
- Type: Article
- + Show details - Hide details
-
p.
217
–220
(4)
The throughput and delay performance of adaptive spreading gain uplink packet transmission in real-time and non-real-time data integrated CDMA networks are analysed under AWGN and Rayleigh fading channels. The processing gain for the non-real-time data is adaptively changed according to the total uplink interference level so that a required signal-to-interference ratio (SIR) can be achieved. The optimum values for the required SIR with regard to maximising the throughput are derived as a function of packet length. The optimised adaptive rate system achieves significantly improved throughput compared to the non-adaptive rate system. In addition, the adaptive rate system achieves almost linear increase in packet delay with traffic density, while the packet delay of the non-adaptive rate system shows an abrupt increase as traffic density increases.
Thought communication and control: a first step using radiotelegraphy
On-demand packet discard scheme for TCP over ATM-UBR service
Base-station network planning including environmental impact control
Satellite constellation design and radio resource management using genetic algorithm
Receiver performance of frequency division multiple access using spectrally overlapping signals
Throughput–delay performance of interference level-adaptive transmission in voice/data integrated CDMA network with variable spreading gain
-
- Author(s): Nigel Linge
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, page: 221 –221
- DOI: 10.1049/ip-com:20040655
- Type: Article
- + Show details - Hide details
-
p.
221
(1)
- Author(s): T. Bayle ; R. Aibara ; K. Nishimura
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 222 –230
- DOI: 10.1049/ip-com:20040616
- Type: Article
- + Show details - Hide details
-
p.
222
–230
(9)
The differentiated services (DiffServ) architecture offers a scalable alternative to provide quality of service (QoS) guarantees for performance-sensitive applications in the Internet. Within the DiffServ framework, however, an efficient traffic scheduling mechanism is a key component to ensure such QoS guarantees. The authors propose a packet scheduling algorithm called enhanced weighted fair queueing (EWFQ), which enables fair bandwidth sharing while supporting tight bounds on end-to-end delay for real-time traffic such as voice over IP (VoIP) in DiffServ networks. EWFQ allows service classes to be created and proportional weights to be assigned to such classes efficiently according to their resource requirements. Results from simulation studies indicate that the mechanism proposed ensures both the required bandwidth fairness and end-to-end network delay bounds according to the specified weight ratios under various traffic and network conditions. The scheme also has lower implementation complexity, along with scalability to accommodate growing traffic flows at the core routers of the high-speed Internet backbone. - Author(s): S.A. Hussain and A. Marshall
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 231 –237
- DOI: 10.1049/ip-com:20040573
- Type: Article
- + Show details - Hide details
-
p.
231
–237
(7)
Active and programmable networks change the functionality of routers and switches by using agents and active packets. The authors present a new packet scheduling scheme called active scheduling to control and maintain QoS parameters in virtual private networks (VPNs) within the confines of adaptive and programmable networks. In active scheduling an agent on the router monitors the accumulated queueing delay for each service. To control and to keep the end-to-end delay within the bounds, the weights for services are adjusted dynamically by agents on the routers spanning the VPN. If there is an increase or decrease in queueing delay of a service, an agent on a downstream router informs the upstream routers to adjust the weights of their queues. This keeps the end-to-end delay of services within the specified bounds and offers better QoS compared with VPNs using static WFQ. An algorithm for active scheduling is described and simulation results are presented and compared with WFQ. - Author(s): S. Zeadally ; R. Wasseem ; I. Raicu
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 238 –242
- DOI: 10.1049/ip-com:20040283
- Type: Article
- + Show details - Hide details
-
p.
238
–242
(5)
The Internet Protocol version 6 (also known as IPv6) has been developed to replace the current IPv4 protocol. Various operating systems running at end-systems now support IPv6 protocol stacks and network infrastructures (hosts, routers) are currently being deployed to support IPv6 features. IPv6 stacks on end-systems constitute an important component in the migration and adoption of IPv6. To investigate the impact of IPv6 on user applications, an empirical evaluation was conducted on the performance of IPv4 and IPv6 protocol stacks on the Linux operating system for TCP and UDP protocols. The IPv6 performance obtained on Linux was compared with the performance of IPv6 of other commodity operating systems, namely Solaris 8 and Windows 2000. The experimental results demonstrate that the IPv6 protocol stack for Linux outperforms IPv6 stacks of these operating systems. - Author(s): K. Ma and K.M. Sim
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 243 –250
- DOI: 10.1049/ip-com:20040281
- Type: Article
- + Show details - Hide details
-
p.
243
–250
(8)
In shortest path (SP) routing, only the shortest path po between a source-and-destination pair is used to route traffic. However, if all data is ‘selfishly’ routed through po, degradation of network performance due to unregulated traffic along po may result. Recent studies show that multipath routing approaches that observe loop-free invariant (LFI) conditions improve network performance. Two new sets of criteria, loose-strain loop-free (LSLF) conditions and simple loose-strain loop-free (SLSLF) conditions, are defined. It is proved that in any given network, PLFIi→j⊆PLSLFi→j and PLFIi→j⊆PSLSLFi→j, where PLFIi→j, PLSLFi→j and PSLSLFi→j are the sets of paths from i to j found under LFI, LSLF and SLSLF conditions and also that any path pi→j∈PLSLFi→j or pi→j∈PSLSLFi→j is loop-free. A series of simulations were performed on many practical network topologies (e.g. APRANET and NSFNET). Favourable empirical results show that LSLF and SLSLF conditions outperform both LFI and SP in terms of average maximum flow and resource utilisation. - Author(s): N.X. Liu and J.S. Baras
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 251 –257
- DOI: 10.1049/ip-com:20040298
- Type: Article
- + Show details - Hide details
-
p.
251
–257
(7)
The long-run queueing performance of a multi-scale TCP traffic model, the HOO model, is analysed. Since the model links the multi-scale behaviour with practical traffic elements and approximates TCP traffic very well, the analysis is expected to provide insights into the physical interpretation of multi-scale traffic and to give useful results for performance prediction. To derive a meaningful solution and avoid the extreme difficulty of an exact analysis, the authors adopt several techniques to track the problem, among which are techniques to establish equivalent processes to the traffic process or the queue content process in two cases, the case of fast flows and the case of slow flows. Quantitative results for the queue tails are obtained in both cases, and a unified form is derived. It indicates that three levels of traffic elements in different time scales, i.e. the connection, the burst and the packet, all affect the asymptotic queueing performance. It shows quantitatively how the connection determines the index of the queue tail, and the burst and the packet contribute to the tail with their averages. Used with simple statistical inferences, the analytical result is shown to predict the queueing performance of real traffic well. - Author(s): V. Friderikos ; A. Mihailovic ; H.A. Aghvami
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 258 –262
- DOI: 10.1049/ip-com:20040574
- Type: Article
- + Show details - Hide details
-
p.
258
–262
(5)
The current modus operandi of different proposed architectures for all-IP mobile and wireless networks is commonly based on the assumption of best-effort routing and per-class (DiffServ) or per-flow (IntServ) treatment of IP traffic. At the same time, emergence and promotion of QoS routing as an embedded network feature seems orthogonal to the deployment of current μ-mobility protocols. The stance to be defended by the authors is that interactions between μ-mobility solutions and different QoS-aware path selection logics (present in QoS routing) do exist and, depending on the solution supported in the scoped domain, non-desirable phenomena may occur degrading the general packet delivery efficiency. The clashing properties of two sets of solutions reflect in the realisation of mechanisms for instalment, maintenance and adaptability of packet forwarding mechanisms to mobile destinations. The impetus of the paper is to unfold the hidden interactions between these two sets of solutions and to throw sharper light on the current vagueness and lack of specificity of deploying QoS routing schemes in a mobile/wireless network arena by proposing different solutions, while at the same time stating that integration of QoS and mobilty is an important challenge for development of future IP networks. - Author(s): S. Zeadally ; F. Siddiqui ; P. Kubher
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 263 –269
- DOI: 10.1049/ip-com:20040284
- Type: Article
- + Show details - Hide details
-
p.
263
–269
(7)
The wide-scale deployment of Intranet and Internet along with improvements in hardware and software technologies are paving the way for emerging applications, many of which involve the delivery of voice, video and images to end-users. One such application that has generated considerable interest recently is voice over IP (VoIP). The impact and feasibility of using VoIP in both Intranet and Internet environments is explored under typical network real-world measurement conditions. The empirical test measurements and results demonstrate the effects of compression, packet size, delays and jitter on the end-to-end delivery performance of voice ultimately delivered to end-users. An analysis is presented of the delay components of the end-to-end delay for VoIP implementation. This analysis identifies areas that could potentially influence the performance of voice when transported over IP-based networks and simultaneously competing with their data traffic. - Author(s): S.H. Mian
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 270 –279
- DOI: 10.1049/ip-com:20040299
- Type: Article
- + Show details - Hide details
-
p.
270
–279
(10)
Streaming video is expected to account for a considerable percentage of traffic on both wired and wireless networks, and is expected to be encoded in a scalable fashion and have a heterogeneous quality. The author discusses a key standard that has introduced such improvements, namely MPEG-4, and outlines a statistical study of predominantly single layer (non-scalable) and temporal/spatial (scalable) MPEG-4 encoded video of differing quality. The results are reported by illustrating the scaling behaviour of such traffic and conclusions are drawn concerning the potential impact on future telecommunication networks. - Author(s): J. Sivarajah ; D.W. Armitage ; N.M. Allinson
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 280 –286
- DOI: 10.1049/ip-com:20040280
- Type: Article
- + Show details - Hide details
-
p.
280
–286
(7)
The increasing growth on the Internet for multimedia streaming poses severe challenges for the dominant transport control protocol TCP, but without a fair demand on resources such network flows threaten the stability of the Internet. A rate-based transport protocol, employing a SIMD window-based congestion mechanism, is described and comparative simulations with other protocols are provided. Rate-based SIMD exhibits clear advantages in terms of TCP-friendliness, smoothness of transmission rates and appropriate responsiveness to changing network conditions. - Author(s): R.V. Prasad ; H.S. Jamadagni ; H.N. Shankar ; H.N. Shankar
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 287 –291
- DOI: 10.1049/ip-com:20040282
- Type: Article
- + Show details - Hide details
-
p.
287
–291
(5)
Voice conferencing is an essential block of any multimedia system used for collaborative work, as voice is shared by all participants. Floor control is mission-critical here and has been investigated by many to ensure fair resource sharing; yet fixing the number of floors has remained an open problem. A conferee (participant in a conference) can speak only after acquiring the floor. To allow impromptu speech, floor allocation must be made for many concurrent speakers. However, too many concurrent speakers degrade voice intelligibility. Therefore, setting an upper bound for the number of streams (floors) that may be mixed is sine qua non for quality conferencing. The problem of setting an upper bound on the number of floors to support concurrent multi-party audio sessions is addressed. A conjecture based on conversational and qualitative analysis is proposed. A pseudo-measure termed ‘loudness number’ used to manage the number of floors is briefly outlined. The implementation at a functional level on Windows© systems has yielded satisfactory performance. - Author(s): Y. Xiang ; Y. Lin ; W.L. Lei ; S.J. Huang
- Source: IEE Proceedings - Communications, Volume 151, Issue 3, p. 292 –295
- DOI: 10.1049/ip-com:20040526
- Type: Article
- + Show details - Hide details
-
p.
292
–295
(4)
A new method of real-time distributed denial of service (DDOS) attack inspection is introduced, based on changes in the characteristic of network self-similarity. Using the real-time RS (R2S) algorithm, simulation experiments with real DDOS attacks and background traffic have been carried out. The results show that DDOS attacks can be detected effectively and precisely under most situations using the proposed method.
Editorial: Internet protocols, technology and applications (VoIP)
Supporting real-time IP traffic with enhanced service classes in DiffServ networks
Provision of quality of service using active scheduling
Comparison of end-system IPv6 protocol stacks
Loose-strain loop-free conditions for multiple path IP routing
Long-run performance analysis of a multi-scale TCP traffic model
Analysis of cross issues between QoS routing and μ-mobility protocols
Voice over IP in Intranet and Internet environments
Analysis of MPEG-4 scalable encoded video
New TCP-friendly, rate-based transport protocol for media streaming applications
Number of floors for a voice-only conference on packet networks – a conjecture
Detecting DDOS attack based on network self-similarity
Most viewed content for this Journal
Article
content/journals/ip-com
Journal
5
Most cited content for this Journal
We currently have no most cited data available for this content.
