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The convolutional coding is a very popular channel coding technique for the major reason of mitigating the probability of having many retransmissions because of difficult (noisy) communication channels. Sequential decoding is a type of convolutional codes that becomes of interest in wireless communication since it affords a decoding time that can be adaptive to channel state. In this study, the authors propose an improved queuing model, using discrete-time semi-Markov chain, which represents a modified packet retransmission policy over previously proposed queuing model. The authors queuing model mainly describes the behaviour of the buffer, which belongs to intermediate hops, when sequential decoding is implemented and concerns also about packets being transmitted over erroneous channels. The authors aim after conducting queuing analysis to find a real mathematical form for the average buffer occupancy as a network performance metric. Although the improved queuing model when incorporating the new policy for retransmission is very complicated, we are finally able to derive an expression for that performance metric considering practical assumptions. They further illustrate how the modified queuing model has a better impact on the end-to-end delay of messages, being transmitted without any extra buffering requirements needed, than the other relevant proposed queuing model. They conduct a simulation study using computer programming with the same assumptions used for queuing analysis to validate their analytical explanations and results. Furthermore, they validate the correctness of their closed-form expression through comparing its results with those obtained from expression related to a queuing model which discarded employing any retransmission policy. [ABSTRACT FROM AUTHOR]

The authors propose a new power saving scheme based on a buffer-threshold and timer in wireless packet networks. In a standard sleep mode operation with a binary exponential algorithm, a mobile station (MS) can begin a wake period if it is aware of the existence of its own downlink packets during a listening interval. In a light traffic environment, this conventional standard operation consumes a considerable amount of power to switch modes (sleep/listen/wake). To solve this problem, a new power saving scheme based on joint control of buffer-threshold and timer (JCBT) is proposed. To see the performance behaviour of the JCBT scheme, the authors consider the total power consumption per unit time and the average packet transfer delay as performance measures. Based on the performance measures, the authors obtain the optimal solution for the proposed JCBT scheme that minimises the total power consumption per unit time with a given delay requirement. [ABSTRACT FROM AUTHOR]

Long-term evolution-advanced (LTE-A), the true 4G technology of third Generation Partnership Project family, has a new feature that aggregates multiple LTE carriers so that users can be served on multiple component carriers (CCs). This characteristic plays an important role to increase data rate multiple times for LTE-A users with corresponding multiple CCs. However, current proposed packet scheduling (PS) algorithms ignore this effect in the effort of providing new solution that increases system performance. The modified proportional fair algorithms proposed in this study can provide optimised algorithm for PS mechanism in LTE-A system as well as other technologies that use multiple carriers. [ABSTRACT FROM AUTHOR]

A modified Alamouti-coded orthogonal frequency division multiplexing (OFDM) scheme is proposed for cooperative systems in non-selective fading environments. Even with the frequency offset between two distributed transmit antennas, the proposed scheme achieves ideal performance and full rate of Alamouti code. By switching subcarriers in the second OFDM symbol of each Alamouti-coded OFDM symbol pair after a simple-phase rotation in the first OFDM symbol, inter-carrier interference terms can be perfectly cancelled after a simple linear combining with the processing overhead of two times down-conversions and discrete Fourier transform (DFT) operations at each OFDM symbol. [ABSTRACT FROM AUTHOR]

How to provide quality of service (QoS) guarantees in the routing and switching devices has become one of the key research topics in the areas of routing and switching technologies. Differentiated services architecture (DiffServ), which forwards packets based on their per-hop behaviours, is known as a promising way for supporting QoS in a high-speed backbone network scenario. Motivated by the load-balanced Birkhoff-von Neumann switch, which has received much attention recently because of its outstanding extensibility performance, in this study, the authors propose a new switch architecture called load-balanced DiffServ support switch (LBDS). LBDS consists of pipeline traffic bandwidth control, priority-based load balancing, and bandwidth-based priority scheduling to implement fast forwarding for expedited forwarding (EF) traffic and assured forwarding for AF classes of traffic. The authors evaluate LBDS by extensive theoretical analysis and comprehensive simulations. As expected, LBDS can provide bandwidth guarantees for both EF and AF traffic. We also show that under burst traffic arrivals, especially when the traffic arrival pattern is non-uniform and the load is above 0.6, LBDS performs much better delay performance than existing DiffServ support schemes. [ABSTRACT FROM AUTHOR]

The existing congestion control algorithms for streaming service experience the performance degradation over IEEE 802.11 multihop network because they mainly focus on the prevention of the buffer overflow at the specific node. However, congestion collapse happens not because of the buffer overflow but because of the serious contention at multiple nodes. As the MAC layer contention becomes serious, the end-to-end delay and the packet loss rate due to collision increase while throughput smoothness is not guaranteed due to fluctuation of the queueing delay at each node. Here, the authors propose a TCP-friendly congestion control algorithm based on the MAC layer contention state to improve streaming service. Specifically, each flow controls its transmission rate based on the estimated contention state which each node notifies to each flow by marking packets. The simulation results show that the proposed algorithm utilises the channel bandwidth more efficiently with providing throughput smoothness compared with the existing algorithms. [ABSTRACT FROM AUTHOR]

This study systematically investigates the long-term prediction of online, real-time variable-bit-rate (VBR) video traffic, which is a key and complicated component of advanced predictive dynamic bandwidth control for future networks such as multimedia satellite networks. As per the time variation, non-linearity and long-range dependence in the VBR video traffic trace, a novel method of feature extraction based on multi-scale decomposition is proposed. After an analysis of the time–frequency distribution characteristics of the video trace, the wavelet packets that have the trait of arbitrary distinction and decomposition are selected. After space partition of wavelet packets, the best wavelet packet basis for feature extraction is picked. Based on the best basis, fast arbitrary multi-scale wavelet packet transform (WPT) can be done and each higher dimension wavelet coefficient matrix can be obtained. Then, wavelet coefficients prediction is done based on normalised least mean square error (NLMS) or least mean-support vector machine (LS-SVM) algorithm. The long-term prediction of the VBR video traffic is obtained through reverse wavelet transforms on the predicted wavelet coefficients. Numerical and simulation results are provided to validate the claims. [ABSTRACT FROM AUTHOR]

Best-effort streamed video playout representing the most popular and cost-effective multimedia application today is unfortunately very sensitive to any fluctuations in packet loss and delay. There are always further problems for the provision of real-time playout applications over wireless links because of the complicated and highly variable nature of the channel and challenging quality issues for adopting the service in global scales, which necessitates adopting a new and effective buffer mechanism. As a solution to this problem, to smooth out the playout for quality viewing and improved overall systems’ performance, in this study the authors propose a new adaptive media playout (AMP) system that makes the most effective use of a packet-delay prediction algorithm. Here, as the prediction method makes decisions based on the delay interdependency of the adjacent packets the differential auto-regression delay prediction model is used for working out the estimation. The proposed algorithm also adopts more refinement changing steps in the adjustment process and takes into accounts any packet losses with deadline constraint. The simulation results show that the proposed method outperforms the AMP-live method in reducing the average buffer delays as well as enhancing package loss rates for both overflow and underflow. [ABSTRACT FROM AUTHOR]

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. [ABSTRACT FROM AUTHOR]

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. [ABSTRACT FROM AUTHOR]

Resilience issue in Ethernet is becoming even more important when Ethernet is becoming more pervasive. The authors propose a local recovery scheme, named Ethernet ultra-fast switching (EUFS), to address this issue. EUFS is based on the multiple spanning tree protocol, which is the basics of both load balancing and fast recovery mechanisms in large-scale Ethernet networks. For each node, the authors construct a pair of protection trees rooted at this node. Normally, the traffic is forwarded along the working trees. When a link or node fails, the immediate upstream node switches the traffic, which intends to pass through the failed link or node, from the working trees to one of the protection trees rooted at itself. EUFS has the following features. First, it achieves an ultra-fast recovery from any single failure in two-connected networks. Second, it supports backward compatibility, because switches implementing EUFS can inter-operate with the legacy Ethernet switches. [ABSTRACT FROM AUTHOR]

The optical migration-capable network with service guarantees hybrid network is a time-divided packet/circuit hybrid network offering absolute performance guarantees and high resource utilisation. Guaranteed service traffic (GST) follows lightpaths receiving absolute priority over statistically multiplexed (SM) traffic. Transmission resources are fully shared by interleaving SM and GST packets, but packet-switched SM packets experience contention, resulting in a packet loss that should be minimised. This article explores a novel bufferless network approach where the performance of the SM class is satisfied by applying packet-level forward error correction (FEC). Two implementation strategies are proposed: RedSM transmitting redundancy packets as SM traffic and RedGST transmitting redundancy packets as low-priority GST traffic. The performance of the schemes is explored analytically and by simulation. For uniform traffic loads, the RedGST scheme shows the best performance of the two, but even at 60% load the RedSM scheme shows packet loss rates (PLRs)close to 10-5. Simulations with non-uniform traffic loads illustrate the importance of selecting the optimal redundancy ratio; the failure to do so results in up to two orders of magnitude increase in PLR for the RedGST scheme. A much higher tolerance for traffic variations is the main benefit for the RedSM scheme. In view of the results, both schemes are attractive alternatives to buffered contention resolution techniques. [ABSTRACT FROM AUTHOR]

Multihop relaying is one of the promising techniques in future generation wireless networks. The adaptive modulation and coding (AMC) mechanisms can be applied in order to increase the spectral efficiency of wireless multihop networks. However, most of these mechanisms concentrate on the physical layer without taking the queuing effects at the data link layer into account, whose performances are overestimated. Therefore the cross-layer analytical framework is presented in analysing the quality-of-service (QoS) performances of the decode-and-forward (DF) relaying wireless networks, where the AMC is employed at the physical layer under the conditions of unsaturated traffic and finite-length queue at the data link layer. Considering the characteristics of DF relaying protocol at the physical layer, the authors first propose modelling a two-hop DF relaying wireless channel with AMC as an equivalent Finite State Markov Chain (FSMC) in queuing analysis. Then, the performances in terms of queuing delay, packet loss rate and average throughput are derived. The numerical results show that the proposed analytical method can be efficiently applied for studying the issues including the relay deployment and the cross-layer design in the multihop relaying networks. [ABSTRACT FROM AUTHOR]

Network intrusion detection systems (NIDS) have become vital components in securing today's computer networks. To be highly effective, NIDS must perform packet inspection of incoming traffic at or near wire speed. Failing to do so will allow malicious packets to sneak through the network undetected, and thus jeopardising network security. Snort is one of the most popular IDS and intrusion prevention system (IPS) applications. Snort is a publicly available open-source NIDS application that typically runs on Linux. In this study, the authors present and discuss the essential software components of Snort and its underlying Linux support architecture. The authors characterise Snort execution and present an analytical queuing model to give insight into understanding the kernel and Snort behaviour as well as to identify key-dominating factors that strongly influence and impact Snort performance. The authors demonstrate that the current default configurations of the packet reception mechanism of the Linux networking subsystem (a.k.a. NAPI) are not suitable for Snort performance and show that the performance of Snort can be improved significantly by tuning certain configuration parameters, specifically by having a small NAPI budge value of 2. The performance is measured in terms of throughput and packet loss. The authors also measure the packet loss encountered at the kernel level as well as the interrupt rate of incoming traffic. Performance was measured when subjecting a PC host running Snort to both normal and malicious traffic, and with different traffic load conditions. [ABSTRACT FROM AUTHOR]

The hierarchical smoothed round robin (HSRR) packet scheduler that improves the performance of the smoothed round robin (SRR) for real-world packet scheduling scenarios is presented. The SRR suffers from an inability to fairly distribute service in highly skewed weight distributions where such distributions often occur in practice. As a result, the transmission behaviour of the SRR can be bursty and non-smooth for the high weighted (or high priority) flows. A novel aggregation strategy is proposed to reshape the sessions' weight distribution that aims to minimise the variance of the high-priority flows' intra-session service times. The aggregation scheme also provides a new capability that supports the provisioning of high-priority service to an arbitrary set of selected flows. The reshaped weight distribution results in guaranteed lower bound in maximum packet delay for the high-priority flows. Furthermore, empirical analyses show that the cumulative average packet delay and jitter of the high-priority flows are significantly reduced from the original SRR and also overall lower for all flows. Along with the enhanced performance in packet delays, the HSRR scheduler attains a worst-case time complexity of O(1). [ABSTRACT FROM AUTHOR]

Mathematical expressions are derived for the received signal to interference plus noise ratio of uplink single carrier (SC) frequency division multiple access (FDMA) multi-user multiple-input multiple-output (MIMO) systems with spatial frequency domain packet scheduling. The scheduler is able to exploit the available multi-user diversity in time, frequency and spatial domains. Our analysis model is confined to 3GPP uplink SC-FDMA transmission in which we specifically investigate multi-user spatial divsion multiplexing MIMO schemes. [ABSTRACT FROM AUTHOR]

IEEE 802.15.4 standard applies the slotted carrier sense multiple access with collision avoidance (CSMA/CA) for its contention mechanism in the contention access period (CAP). In CAP, channel utilisation is significantly affected by backoff time and collision. A delayed backoff algorithm (DBA) in CAP is proposed for the beacon-enabled networks. DBA improves the goodput and power consumption by decreasing the backoff time between packets and assuring that the different nodes will not finish their backoffs at the same time. Simulation results show that the proposed algorithm outperforms the slotted CSMA/CA and other schemes in terms of goodput and power consumption. [ABSTRACT FROM AUTHOR]

The performance of direct-sequence code division multiple access (DS-CDMA) using space–time spreading system, over frequency-selective fading channels, is investigated. The underlying transmit diversity scheme, previously introduced in the literature, is based on two transmit and one receive antenna. It was shown that when employed in flat fast-fading channels, the received signal quality can be improved by utilising the spatial and temporal diversities at the receiver side. We study the problem of multiuser interference in asynchronous CDMA systems that employ transmit/receive diversity using space–time spreading. To overcome the effects of interference, a decorrelator detector is used at the base station. Considering binary phase-shift keying transmission, we analyse the system performance in terms of its probability of bit error. In particular, we derive the probability of error over frequency-selective Rayleigh fading channels for both fast and slow-fading channels. For the fast-fading channel, both simulations and analytical results show that the full system diversity is achieved. On the other hand, when considering a slow-fading channel, we show that the scheme reduces to conventional space–time spreading schemes where the diversity order is half of that of fast-fading. [ABSTRACT FROM AUTHOR]

The IEEE 802.11e standard is specified to support quality-of-service in wireless local area networks, and different contention parameters are designated to each type of service. This developed model is presented to analyse the scheduling and the contention between packets with different priorities, where the new features of the enhanced distributed channel access such as virtual collision, backoff, minimum contention window and different arbitration inter-frame spaces are taken into account. Based on the model, the delay performance of differentiated service traffic is analysed and a recursive method is proposed, which is capable of calculating the mean access delay. Simulations show that the model and the analysis provide an insight into the protocol and the effects of different parameters on the performance. [ABSTRACT FROM AUTHOR]

An optimisation framework for wireless link adaptation which maximises the system throughput subject to a packet-level quality of service (QoS) constraint is presented. The authors consider joint adaptive variation of the transmitted power level, transmission data rate and packet-error-rate (PER) at the physical layer to improve the throughput performance of the selective-repeat automatic repeat request (SR-ARQ) protocol over block-fading channels. Specifically, for discrete-rate coded M-QAM schemes, the authors present a power and rate adaptation algorithm that guarantees a target PER constraint. The proposed framework also facilitates optimising the throughput performance for delay constrained wireless applications, which imposes a limit on the number of retransmissions for the ARQ. In particular, a link adaptation scheme is presented, which guarantees a target packet-loss rate (PLR) when a truncated SR ARQ protocol at the data-link layer is employed. Numerical results indicate that the proposed adaptation schemes compared with other adaptive schemes noticeably enhance the system throughput. Also, in the case of truncated ARQ, the results illustrate a fundamental trade-off between the delay and PLR QoS metrics achieved at the maximum throughput performance. [ABSTRACT FROM AUTHOR]

The authors review analysis, simulation and measurement techniques, the three fundamental methods for performance evaluation in packet networks, looking at what's known, what's new and some outstanding issues. In trying to avoid re-reviewing material which has already been well summarised elsewhere, the authors concentrate on areas that are relatively new or possibly less generally well appreciated. So, under analysis, the focus is on models for network topologies and connectivity, and on wireless access. In the simulation section the focus is on techniques for scalable simulation for large-scale packet networks. Compared to the other two areas measurement is relatively new anyway, and more time is spent on motivation, techniques and some recently discovered limitations. [ABSTRACT FROM AUTHOR]

The article discusses various reports published within the issue, including "Analysis, Simulation and Measurements in Large-scale Packet Networks," "Adaptive Wavelet Based Multifractal Model Applied to the Effective Bandwidth Estimation of Network Traffic Flows," and "A Precedence-Enabled Per Hop Behavior: Impact on TCP and UDP Flows."

Packet-level measurement is now critical to many aspects of broadband networking, for example for guaranteeing service level agreements, facilitating measurement-based admission control algorithms and performing network tomography. Because it is often impossible to measure all the data passing across a network, the most widely used method of measurement works by injecting probe packets. The probes provide samples of the packet loss and delay, and from these samples the loss and delay performance of the traffic as a whole can be deduced. However, measuring performance like this is prone to errors. Recent work has shown that some of these errors are minimised by using a gamma renewal process as the optimal pattern for the time instants at which to inject probes. This leaves the best rate at which to inject probes as the key unsolved problem, and this is addressed here by using the statistical principles of the design of experiments. The experimental design approach allows one to treat packet-level measurements as numerical experiments that can be designed optimally. Modelling the overflow of buffers as a 2-state Markov chain, the system's likelihood function is deduced, and from this a technique (using the Fisher information matrix) to determine the upper-bound on the optimal rate of probing is developed. A generalisation of this method accounts for the effect of the probed observations interfering with the experiment. The numerical results focus on VoIP traffic, allowing one to show how this methodology would be used in practice. One application of this is in measurement-based admission control algorithms, where the technique can be used to provide an upper-bound on the rate at which probes should be injected to monitor the loss performance of the target network, prior to making an admit/do not admit decision. [ABSTRACT FROM AUTHOR]

Packet scheduling is a vital component to support different classes of service in all-packet networks. In classical queuing systems, the waiting-time performance of non-first in first out buffer scheduling systems could be predicted through the use of analysis. However, all-packet networks feature traffic patterns that do not conform to classical Poisson-like processes, and this greatly complicates the evaluation of their performance. Our novel approach to this problem is through a hybrid combination of analysis and simulation. The authors derive a combinatorial algorithm, using the generalised ballot theorem, which predicts waiting times for low-priority traffic. When this algorithm is combined with prior work on traffic aggregation, the authors achieve a significant reduction in the state space associated with the buffer under study. To numerically test this algorithm, the authors demonstrate its use in simulation, where state space and event count reduction is a fundamental requirement to ensure experiments complete in a timely fashion. Numerical results from these simulations show a very significant reduction in the number of events processed combined with improved state coverage. This is achieved while maintaining a highly accurate representation of packet delays compared with a conventional approach. [ABSTRACT FROM AUTHOR]

The author proposes and analyses an approach to support both precedence and preemption (P&P) and QoS handling in common packet transport infrastructures. Precedence has to do with the relative importance of the information content, whereas preemption has to do with mechanisms to deny lower precedence traffic access to network resources in favour of higher precedence traffic, when necessary. The author's approach to this duality is to enhance active queue management (AQM) techniques to provide P&P capabilities and rely on standard, well-studied QoS schedulers, for example, Weighted Round Robin, Class-Based Weighted Fair Queuing, etc., for handling QoS requirements. The present scheme allows low-order queues (within the context of QoS handling) to plead up to the next higher-order queue for help in alleviating queue congestion under periods of communication link overload. The scheme is referred as the cross-queue-AQM (CQ-ACM) scheme and can be extended to higher numbers of queues and any type of scheduler. Through extensive simulation studies, the performance of CQ-AQM scheme under heavy traffic limits is investigated, where preemption is required. The performance metrics of interest to this analysis are packet delay, packet loss and throughput as a function of the packet QoS class and P-L. Further, two new metrics of merit, Gain and System Efficiency, are defined, which are specific to P&P considerations. The studies concentrated on both flow-controlled and non-flow-controlled traffic, as well as mixed traffic conditions. Simulation results show that the algorithms perform extremely well for all cases studied. [ABSTRACT FROM AUTHOR]

The provision of quality of service to different service types in broadband packet networks requires techniques for the evaluation of the probabilities of packet delays and loss. The authors analyse the multiplexing of independent and homogeneous Markovian ON–OFF traffic sources into a single buffer, extending earlier work. The resulting closed-form equation provides a virtually zero-complexity approach to the calculation of the buffer overflow probability via the burst-scale decay rate of the buffer state probabilities. Graphical results are provided, comparing our expression with the results from simulations and the ‘standard’ formula of Anick, Mitra and Sondhi. These show that our new formula provides excellent accuracy for medium to high loads, i.e. those load values at which queuing becomes important. This makes it ideal for quick calculations, with practical utility when dimensioning checks are required in many networking situations involving the accurate estimation of loss probabilities in e.g. buffer multiplexing VoIP traffic sources. [ABSTRACT FROM AUTHOR]

Loss-free schemes are defined to ensure successful packet/burst transmissions in optical packet/burst switching networks. To this end, they rely on a collision-free routing and wavelength assignment (CF-RWA) scheme combined with simple contention resolution mechanisms that guarantee the absence of losses in intermediate links. Here, the CF-RWA problem is studied. In particular, by using graph theory, the problem of finding CF-RWA schemes that minimise the number of wavelengths to serve a given traffic matrix is set. The problem is simplified when it is formulated by using pre-defined sets of non-colliding paths. Within this framework, the problem is shown to be equivalent to finding a given vertex-set colouring of the so-called restriction digraph. Here, two heuristic algorithms are proposed to obtain such vertex-set colourings. One of them provides a suitable CF-RWA without having to solve the minimisation problem. By way of example, the proposed method is applied to the NSFNet and the EON network providing quasi-optimal results. [ABSTRACT FROM AUTHOR]

First, the optical burst switching (OBS) network with limited wavelength conversion capability (LWCC) is decomposed into multiple wavelength continuous segments according to the position of wavelength converters. Based on the decomposed network model, two reservation signalling mechanisms for OBS-LWCC networks, namely wavelength-amend-on-demand (WAoD) and contention-based limited signalling protocol (CLSP), are proposed to reduce the blocking probability in the OBS-LWCC networks. In WAoD, the congested node sends a feedback message to the closest upstream switch with wavelength conversion capability. The notified switch will change the burst wavelength to the one assigned by the congested node to avoid collision in advance. Extensive simulation results indicate that by applying the wavelength reconfiguration method to the core nodes without wavelength conversion capability, the proposed WAoD scheme can improve the blocking performance in the OBS-LWCC networks significantly. Furthermore, CLSP, which combines the burst time-slot reconfiguration with the wavelength reconfiguration of WAoD, always has the lowest burst loss probability when compared with both the conventional and the proposed reservation mechanisms. [ABSTRACT FROM AUTHOR]

A protection and restoration mechanism is essential to provide reliable traffic transmission in networks, but it is not easy to apply existing protection mechanisms to optical burst-switching (OBS) networks, because of the unique properties of OBS. The authors introduce a 1:1 link-based protection that minimises burst losses by deflecting bursts until a source ingress router arbitrates a working path to a backup path when a link failure occurs. The authors also propose a genuine dynamic resource sharing (DRS) link protection in order to achieve high efficiency in network resource allocation and reliable protection simultaneously in OBS networks. DRS employs burst-multiplexing and label-stacking techniques in backup channel selection and always reserves an optimal number of backup channels according to the changes in traffic load in a working link and the quality of service (QoS) requirements of bursts. Simulation results show that the DRS can provide the same protection reliability with just half of the backup channels needed in a normal 1:1 protection in a light-load case, while guaranteeing the QoS requirements of class bursts. The authors show that this property also makes backup link sharing among a number of working links possible. [ABSTRACT FROM AUTHOR]

A strategy for minimising the probability of contention in optical burst-switching (OBS) networks, by combining traffic engineering in the wavelength domain with delayed burst scheduling at the ingress nodes, has been proposed. The implementation of this strategy only requires an increase on the processing and electronic buffering capacities of the ingress nodes, while avoiding the use of complex and expensive optical buffers at the core nodes. The performance evaluation study has shown that the proposed strategy outperforms the burst-scheduling strategies described in the literature by several orders of magnitude, in terms of burst-blocking probability. Moreover, the performance improvements have been shown to depend on the average offered traffic load, number of wavelengths per link and maximum delay allowed at the ingress nodes. [ABSTRACT FROM AUTHOR]

Over the past several years, a significant amount of research has been conducted in the areas of optical packet switching (OPS) and optical burst switching (OBS). This research has been motivated by the need for techniques that are capable of supporting the demanding requirements of emerging dynamic high-bandwidth network applications in a flexible and efficient manner. Although optical packet and burst switching have yet to be widely deployed in commercial settings, recent research progress indicates that such deployments are not infeasible in the near future. The authors review the literature on OPS and OBS. Basic concepts are discussed and an overview of current and emerging research issues and challenges for optical packet and burst switched networks is presented. The authors attempt to cover issues that are essential for the practical deployment of such networks and highlight promising research directions that are likely to facilitate such deployments. [ABSTRACT FROM AUTHOR]

A new scheduling algorithm called serialised batch scheduling (SBS) for optical burst switching (OBS) networks is proposed. SBS aggregates and serialises bursts along a shared path into one composite burst, which is switched as one unit. There are two major processes in SBS, namely, batching and serialising. While the batching process chooses a set of bursts to form the composite burst, the serialising process determines how to organise the OBS bursts within the composite burst and generates a corresponding control packet for this burst. Several SBS batching and serialising schemes are introduced and analysed here. The study by the authors shows that the guard band and burst overlap can be reduced in the SBS and, thus, the packet loss rate and the number of switch reconfigurations can be significantly reduced. In addition, it is indicated that the proposed SBS algorithm can be coupled with other OBS scheduling algorithms and reduce the requirements for a high-speed optical switch in OBS networks. [ABSTRACT FROM AUTHOR]

In future high-speed photonic packet switching networks, it is highly desirable to carry out robust all-optical header recognition to provide high-throughput routing. The authors present a pulse-position modulation header processing (PPM-HP) scheme, offering significantly reduced routing table size and employing a single bit-wise AND operation to carry out correlation of the packet header with the entire routing table entries. The downsized routing table also offers multiple transmission modes (unicast, multicast and broadcast) in the optical layer and improves core network scalability where the number of core/edge nodes could be altered without the need for changing the number of routing table entries. The authors present modelling and simulation of the packet switching router based on PPM-HP. Noise propagation and crosstalk incurred in a multiple-hop routing scenario are investigated. The simulation results are presented and compared with the theoretical calculations. [ABSTRACT FROM AUTHOR]

The author evaluates the performance of bufferless optical packet switches equipped with shared tuneable optical wavelength converters (TOWC) and supporting the quality of service (QoS). The TOWCs are shared per output fibre and, hence each output fibre has a dedicated pool of TOWCs. The TOWCs are not completely shared within the node. The QoS, expressed in terms of the packet loss probability, is differentiated by giving the packets different priorities in accessing both the output wavelengths and the TOWCs. A scheduling algorithm is proposed and its complexity is evaluated. The QoS technique's effectiveness is studied by means of an analytical model based on a probabilistic approach and validated by simulation results. The results show that the proposed technique to support the QoS facilitates a good differentiation of the QoS offered to packets. [ABSTRACT FROM AUTHOR]

Most of the previous works on optical burst switching (OBS) assume in their analysis that signalling does not affect network performance. It is analysed here, under which conditions the effect of signalling is actually negligible, taking into account the effect of signalling in the evaluation of burst discard probability. First, analytical models for two different signalling approaches in an OBS network are presented: ‘out-of-band’ and ‘in-band’ techniques. The impact of these two signalling strategies in terms of the probability of burst discard are evaluated, identifying the component of bursts discarded as a consequence of control message losses or of excessive signalling delay. A new method is also discussed, based on the previous models, to assign the correct amount of resources to the control plane. To verify the accuracy of the analytical results, these are compared with results based on discrete-event simulationns: results are found to be in a highly satisfactory agreement with simulations. [ABSTRACT FROM AUTHOR]

Future interconnection networks will be required to achieve ultra-high bandwidth and low latency communications to cope with the increasing performance requirements of backbone routers, large data storage systems and supercomputing systems. Aiming at achieving ultra-high bandwidth communications and approaching optical time-of-flight processing latency while being robust to cascade impairments, the authors propose an all-optical packet-switched interconnection network, where not only the actual packet switching but also the packet processing is performed in the photonic domain. The authors present two modular architectures, based on the crossbar and the Batcher–Banyan topologies, capable of forwarding fixed-length packets with two classes of service. Both use photonic digital-processing subsystems built by combining a single integrable module which exploits cross gain modulation in a semiconductor optical amplifier. System level simulations on the crossbar switch controller guarantee that the control signals maintain an acceptable quality during the processing. Moreover, the Batcher–Banyan configuration is more cost-effective than the crossbar for increasing port count, while effective network performance in terms of packet loss rate can be obtained by adding just few recirculating delay lines. [ABSTRACT FROM AUTHOR]

A packet switch architecture and a method for load-balancing are described, which involve no centralised schedulers. The method is implemented by distributing extended cross-point queues (a three-dimensional structure) over all elements of the switch and deploying pollers to append packets and to select the queues to be served, together with simple local work-conserving schedulers. The queue structure is such that it renders the proof that no packet will be mis-sequenced trivial. The architecture is practical and shows enhanced performance compared with other state-of-the-art load-balancing architectures, not only for the average delay but also for the distribution of individual delays, the latter being measured by a custom tool that compares the performance of the architecture to the ideal operation of an output queued switch. The queue structure permits the fair penalisation of only the offending input–output flows within the switch in the case of buffer overflow. The basic scheme is enhanced to avoid improper operation in the presence of packet drops, a problem that reintroduces mis-sequencing and that has not been properly addressed in the class of architectures that use pollers to distribute packets. [ABSTRACT FROM AUTHOR]

Aiming to generate ultra wideband (UWB) pulses based on wavelet packets, four different circuits are proposed. It has been shown that the wavelet packets are quasi-orthogonal and have almost identical time duration, and therefore appear to be valuable candidates for UWB pulse shape modulation systems. However, as with any technology, circuitry is an important challenge to solve before a theoretical application can live up to its full potential. The four proposed circuits are based on different interpretations of the equation defining the wavelet packets and their performance is investigated by computer simulation. The research also illustrates the characteristics and the consequent implementation advantages or issues afferent to each circuit. [ABSTRACT FROM AUTHOR]

It is desirable to construct a transmission matrix with full code rate as well as full transmit diversity in the design of space time block coding (STBC). For an arbitrary complex constellation, only full rate and full diversity signal matrix with size 2 by 2 is possible, said as Alamouti Code. For a real signal matrix with a size above 8 by 8, it is also impossible to acquire full code rate and full transmit diversity simultaneously. An efficient selective receiver switching scheme is proposed for STBC with the full code rate and non-orthogonal design. In the proposed scheme with the aid of beamforming, the authors divide the received signals into two groups according to the encoded matrix. By this way, the authors can eliminate the interference from the neighbouring signals almost by half. The simulation results with the example of matrix demonstrate that the proposed scheme can provide the improved performance and more under imperfect channel estimation. [ABSTRACT FROM AUTHOR]

Researches and development efforts in wireless networking and systems are progressing at an incredible rate. Among them, measurement and analysis of performance achieved at network layer and perceived by end users is an important task. In particular, recent advances concerning IEEE 802.11b-based networks seem to be focused on the measurement of key parameters at different protocol levels in a cross-layered fashion, because of their inherent vulnerability to in-channel interference. By adopting a cross-layer approach on a real network set-up operating in a suitable experimental testbed, packet loss against signal-to-interference ratio in IEEE 802.11b-based networks is hereinafter assessed. Results of several measurements aimed at establishing the sensitivity of IEEE 802.11b carrier sensing mechanisms to continuous interfering signals and evaluating the effects of triggered interference on packet transmission. [ABSTRACT FROM AUTHOR]

The incremental redundancy mechanism defined for enhanced general packet radio service (EGPRS) (also known as type-II hybrid ARQ) is considered here. The optimum packet combining strategy is identified and is then used to assess the performance of transmission control protocol/internet protocol in the transmission of bulk data over an EGPRS system. The influence of other key system operating parameters is also assessed and a set of optimal operating parameters is derived and discussed. [ABSTRACT FROM AUTHOR]

Directional distributed co-ordination function (D-DCF), a modified medium access control protocol of IEEE 802.11, is proposed to support hybrid antennas such as smart adaptive array antennas and normal omni-directional antennas, in one wireless local area network cell. Nodes equipped with smart antennas follow D-DCF and nodes equipped with normal antennas follow DCF. D-DCF based on a hybrid virtual carrier sense mechanism maintains compatibility with DCF. In D-DCF, before sending any data-frames, the sender and receiver node transmit a pilot sequence by means of an omni-directional request-to-send/clear-to-send handshake mechanism. Based on the pilot, the directional beam can be formed by the smart antenna. Then the node can transmit its data-frame in the directional mode. The other nodes save the transmission time between the sender and receiver in the omni-directional mode in their network allocation vectors. When the sender and receiver communicate in the directional mode, the other nodes can access the channel to send their data-frames. Hence, D-DCF supports space division multiplexing. Moreover, D-DCF fully supports time division duplex, namely both the forward and backward transmission in one access period. Simulation results show that D-DCF can support the hybrid antenna system effectively and provide much higher network throughput, lower delay, jitter and packet-loss-rate than DCF does. [ABSTRACT FROM AUTHOR]

An analytical approach based on a reduced Markov chain (RMC) model of an optical packet switch with partially shared output buffer has been presented and a combined scheme for the contention resolution in optical packet switches has been proposed. The proposed scheme takes the advantage of output buffering, shared buffering and wavelength conversion to enhance the switch performance. The performance evaluation of the switch architecture based on RMC modelling has been validated through extensive simulation. Moreover, for a self-similar input traffic, the switch performance has been shown to achieve further improvement if a traffic shaping mechanism is incorporated at the input end of the switch architecture. [ABSTRACT FROM AUTHOR]

In this paper we study the accuracy achievable when using probing to measure packet loss probability. Active monitoring using packet probing is widely used to measure network performance at the packet level, especially where direct measurements from network nodes is not possible. In this paper we develop an analytical solution that allows us to predict the number of samples (probes) needed to measure packet loss probability. This formula relates the number of probes needed to the networking scenario, the traffic characteristics, the target packet loss probability and the accuracy to which this loss probability must be resolved. We show that the number of probes required to accurately measure the packet loss probability is a highly sensitive function of the link bandwidth, the traffic load and burstiness, and the packet loss probability itself. We quantify how many probes are needed in certain typical scenarios involving either VoIP or generic data traffic models. [ABSTRACT FROM AUTHOR]

The article introduces several papers discussed within this issue, including one on the amount of research that has been conducted in optical burst switching (OBS) and optical packet switching (OPS), and another on scheduling.

An ant-based contention resolution scheme for the slotted optical packet switched networks has been proposed, and the optimal number of fibre delay lines (FDLs) used in the switch under various traffic loads is investigated. The proposed algorithm can makes FDLs and output port assignment so as to improve the cell-loss rate under various traffic loads. It is found for traffic loads 0.9, 0.6, 0.3, the optimum value for FDL is 64 128 and 128. [ABSTRACT FROM AUTHOR]