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Both economic and environmental costs are driving much research in the area of the energy efficiency of networking equipment. This research has produced a great amount of proposals. However, the majority of them remain unimplemented due to the lack of flexibility of current hardware devices and a certain lack of enthusiasm from commercial vendors. At the same time, Software-Defined Networking (SDN) has allowed customers to control switching decisions with a flexibility and precision previously unheard of. This paper explores the potential convergence between the two aforementioned trends and presents a promising power saving algorithm that can be implemented using standard SDN capabilities of current switches, reducing operation costs on both data centers and wired access networks. In particular, we focus on minimizing the energy consumption in bundles of energy-efficient Ethernet links leveraging SDN. For this, we build on an existing theoretical algorithm and adapt it for implementing with an SDN solution. We study several approaches and compare the resulting algorithms not only according to their energy efficiency, but also taking into account additional QoS metrics. The results show that the resulting algorithm is able to closely match the theoretical results, even when taking into account the requirements of delay-sensitive traffic.

To solve the problem between the high QoS requirement and bandwidth impartiality when the real-time traffic transmitted in IP network,load-adaptive slotted priority queue（LASPQ） was introduced. According to the load of the real-time traffic,it was transferred with an appropriate triggering period τ. LASPQ keep real-time traffic’s delay,jitter in need,while improving the transferring efficiency of non-real-time traffic. It is proven from theory and simulation that LASPQ is better than traditional queue schemes such as PQ,WFQ,when transmitting real-time traffic.

The emerging vehicular ad hoc network (VANET) enables vehicles to access the Internet through roadside access points (APs). An important issue in system deployment is determining how many roadside APs shall be installed on a road. However, the existing works fail to provide rigorous and accurate analysis for VANETs. In this paper, we propose a general structure for Internet access in VANETs. It allows both real-time traffic and delay-tolerant traffic to be delivered to users in the most efficient ways. An analytical model is also proposed to analyze the system performance with random arrival of the vehicles. We finally develop an AP placement algorithm based on theoretical results derived from the model to deploy the minimal number of roadside APs with quality-of-service (QoS) guarantees. The simulation results have demonstrated the accuracy of the proposed analytical model and the efficiency of the proposed algorithm. [ABSTRACT FROM PUBLISHER]

Motivated by the regular service requirements of video applications for improving quality of experience (QoE) of users, we consider the design of scheduling strategies in multihop wireless networks that not only maximize system throughput but also provide regular interservice times for all links. Since the service regularity of links is related to the higher-order statistics of the arrival process and the policy operation, it is challenging to characterize and analyze directly. We overcome this obstacle by introducing a new quantity, namely the time-since-last-service (TSLS), which tracks the time since the last service. By combining it with the queue length in the weight, we propose a novel maximum-weight-type scheduling policy, called Regular Service Guarantee (RSG) Algorithm. The unique evolution of the TSLS counter poses significant challenges for the analysis of the RSG Algorithm. To tackle these challenges, we first propose a novel Lyapunov function to show the throughput optimality of the RSG Algorithm. Then, we prove that the RSG Algorithm can provide service regularity guarantees by using the Lyapunov-drift-based analysis of the steady-state behavior of the stochastic processes. In particular, our algorithm can achieve a degree of service regularity within a factor of a fundamental lower bound we derive. This factor is a function of the system statistics and design parameters and can be as low as two in some special networks. Our results, both analytical and numerical, exhibit significant service regularity improvements over the traditional throughput-optimal policies, which reveals the importance of incorporating the metric of time-since-last-service into the scheduling policy for providing regulated service. [ABSTRACT FROM AUTHOR]

A recent trend in Internet usage has seen large amounts of multimedia data due to increasingly large numbers of mobile users. To facilitate higher bandwidth, modern mobile routers are capable of supporting simultaneous multi-band, leading to less interference, higher capacity and better reliability. However, there exists neither previous work that attempts to maximize utilization of available bandwidth in order to increase performance of multi-band mobile routers through the sharing of traffic classes among different frequency bands of the multi-band mobile router by scheduling, nor that shows multi-band routers are better than single band routers in realistic scenarios. In this paper, we have proposed a novel scheduling algorithm for multi-band mobile routers which transmits different classes of traffic through different frequency bands to achieve improved performance. We have developed an analytical model to perform queuing analysis of the proposed scheduling algorithm for multi-band mobile routers and derived various performance metrics, validated by extensive simulations. Our results show that the proposed scheduling model can improve performance by ensuring maximum possible utilization through sharing of capacities among the bands in multi-band systems. We show our results by comparing proposed multi-band scheduling model with single and current multi-band scheduling models. In addition, we compare single and multi-band mobile routers. It is evident from our results that multi-band systems are not always better than single band systems although multi-band systems are expected to have better performance. Based on the results, we have listed recommendations for selecting single or multi-band systems and allocation policies according to traffic conditions, and their priorities. Our proposed scheduling algorithms and related analysis will help network engineers build next generation mobile routers with higher throughput and utilization, ensuring less packet loss of different classes of traffic. [ABSTRACT FROM AUTHOR]

In this paper, we propose a QoS-aware dynamic bandwidth allocation scheme to mitigate congestion problem in gateway-based multi-hop WiFi-based long distance networks and thereby enhance QoS guarantees for real-time traffic. It presents a dynamic slot scheduling scheme which efficiently distributes the unused TDMA time slots among the needy nodes. The distribution process is carried out in a hierarchical manner through the use of parent-child relationship of a tree topology. By doing so, end-to-end performance of real-time traffic is enhanced manifold. Furthermore, the proposed protocol provides assured packet forwarding for already admitted flows by allowing transmission of nodes according to the capacity of ancestral links. Simulation results show that the proposed protocol achieves a significant performance improvement in terms of throughput and delay of real-time traffic. [ABSTRACT FROM AUTHOR]

Scheduling is the method of dynamically allocating radio resources to user equipment based on scheduling algorithms involved in the long term evolution (LTE) base station. Several algorithms have been proposed by network researchers as the realization of scheduling algorithm which explains an open issue in LTE standard to be able to meet the quality of service requirements for different applications. In this paper, we evaluate the performance of six scheduling algorithms suggested for LTE downlink transmission. The analysis is inspected for a multicellular with interference scenario for real-time and non-real-time traffic. [ABSTRACT FROM AUTHOR]

In this paper the problem of Quality of Service (QoS) provisioning for the Real-Time secondary traffic in Cognitive Radio Networks (CRNs) is investigated. Supporting the Real-Time (RT) traffic in CRNs is a complicated task which requires elaborate consideration in different aspects of the network. Here we consider a practical model for the network. Furthermore, a novel packet scheduling scheme is proposed which can efficiently provide the QoS requirements of the RT Secondary Users (SUs). The variations of the SNR of wireless frequency channels which result in variable transmission rates on the channels are also taken into account in the proposed scheme. The resulting QoS parameters of the RT SUs are calculated analytically. The obtained results are also verified by simulations. The results provide helpful insight for the support of RT traffic in CRNs. [ABSTRACT FROM PUBLISHER]

In this paper, we evaluate the schedulability of traffic with arbitrary end-to-end deadline constraints in Wireless Mesh Networks (WMNs). We formulate the problem as a mixed integer linear optimization problem, and show that, depending on the flow aggregation policy used in the network, the problem can be either convex or non-convex. We optimally solve the problem in both cases, and prove that the schedulability does depend on the aggregation policy. This allows us to derive rules of thumb to identify which policy improves the schedulability with a given traffic. Furthermore, we propose a heuristic solution strategy that allows good suboptimal solutions to the scheduling problem to be computed in relatively small times, comparable to those required for online admission control in relatively large WMNs. [ABSTRACT FROM AUTHOR]

In the last few years wireless local area network (WLAN) technology and products have grown unusually. This growth has encouraged many researchers to employ WLAN technology in different application Areas. Recently, using IEEE 802.11 WLAN in industrial automation applications is attractive and more interesting. However, the currently used medium access mechanism makes WLANs nondeterministic and hence not suitable for industrial real-time traffic. Legacy IEEE 802.11 WLANs were initially designed for best effort traffic only and did not provide any QoS support for this kind of traffic. The new amendment IEEE 802.11e standard was introduced and ratified in 2005. It defines the concept of a Hybrid Co-ordination Function (HCF) at the MAC layer for medium access control. HCF is a combination of HCF Controlled Channel Access (HCCA) with parameterized quality of service (QoS) and Enhanced Distribution Channel Access (EDCA) with prioritized QoS. Enhanced Distribution Channel Access (EDCA) is an enhanced version of Distributed Coordination Function )DCF( mechanism used in the original IEEE 802.11 WLAN. This paper analyze the performance of the EDCA in an industrial automation network with real-time requirements by means of a simulation with the network simulator OPNET 11.5 and compares the results with the DCF in terms of latency and throughput in various scenarios.

Applications of video streaming and real-time gaming, which generate large amounts of real-time traffic in the network, are expected to gain considerable popularity in Long Term Evolution networks. Maintaining the QoS such as packet delay, packet loss ratio, median, and cell border throughput requirements in networks dominated by real time traffic, is critical. The existing dimensioning methodology does not consider QoS parameters of real-time traffic in network dimensioning. Moreover, exhaustive and time-consuming simulations are normally required to evaluate the performance and QoS of real-time services. To overcome this problem, we propose an improved radio network dimensioning framework that considers the QoS of real-time traffic in network dimensioning. In this framework, an analytical model is proposed to evaluate the capacity and performance of real-time traffic dominant Long Term Evolution networks. The proposed framework provides a fast and accurate means of finding the trade-off between system load, packet delay, packet loss ratio, required median, and cell border throughput. It also provides network operators with an analytical means for obtaining the minimum number of sites required by jointly considering coverage, capacity and QoS requirements. The accuracy of the proposed model is validated through simulations. [ABSTRACT FROM AUTHOR]

Highlights: [•] Transceivers’ traffic flow(s) are adjusted for prevailing network conditions. [•] Traffic is asynchronously off-loaded if control actions are not enough. [•] Delay and change of delay are used for packet size definition. [•] Needs for control actions and off-loading are automatically detected. [•] Number of concurrent users are increased and quality of the connections improved. [ABSTRACT FROM AUTHOR]

In IEEE 802.16 specifications, a CRC field can be attached in the rear of a WiMAX MPDU. Thus, each intermediate node (relay node or base station) in an 802.16-based multihop network can perform error detection on a received WiMAX MPDU. On the sender, the ARQ mechanism will be enabled to retransmit the MPDU if a transmission error occurs and an ACK message cannot be received. However, the ARQ mechanism is unsuitable for real-time traffic due to the delay consideration and the CRC field becomes an option to reduce the overhead of a WiMAX MPDU. Once the CRC field is removed from a WiMAX MPDU, the error occurrence cannot be detected by the intermediate nodes in time, and the erroneous data is still forwarded in the network but discarded by the destination. The resources of nodes in a network are wasted. Hence, the error detection mechanism should be retained for real-time traffic, but the overhead of CRC should be reduced. In this paper, we study an efficient discrete-error-checking scheme (DECS) to reduce the CRC overhead for real-time traffic. The DECS only needs several CRC checking bits, which are attached behind some WiMAX MPDUs. The DECS can efficiently provide the error detection for the intermediate nodes to discard these erroneous and useless data early. Thus, the utilization of node resources and wireless bandwidth can be improved significantly. [ABSTRACT FROM AUTHOR]

The proliferation of different wireless access technologies, together with the growing number of multi-radio wireless devices suggest that the opportunistic utilization of multiple connections at the users can be an effective solution to the phenomenal growth of traffic demand in wireless networks. In this paper, we consider the downlink of a wireless network with N access points (\ssr APs) and M clients, where each client is connected to several out-of-band \ssr APs, and requests delay-sensitive traffic (e.g., real-time video). We adopt the framework of Hou and study the maximum total timely throughput of the network, denoted by C{\ssr T^{3}}, which is the maximum average number of packets delivered successfully before their deadline. Solving this problem is challenging since even the number of different ways of assigning packets to the \ssr APs is N^M. We overcome the challenge by proposing a deterministic relaxation of the problem, which converts the problem to a network with deterministic delays in each link. We show that the additive gap between the capacity of the relaxed problem, denoted by Cdet and C{\ssr T^{3}} is bounded by 2\sqrtN(Cdet+N\over4), which is asymptotically negligible compared to Cdet, when the network is operating at high-throughput regime. In addition, our numerical results show that the actual gap between C{\ssr T^{3}} and Cdet is in most cases much less than the worst-case gap proven analytically. Moreover, using LP rounding methods we prove that the relaxed problem can be approximated within additive gap of N. We extend the analytical results to the case of time-varying channel states, real-time traffic, prioritized traffic, and optimal online policies. Finally, we generalize the model for deterministic relaxation to consider fading, rate adaptation, and multiple simultaneous transmissions. [ABSTRACT FROM PUBLISHER]

Rail systems are highly complex and their control requires mathematical-computational tools. The main drawback of the models used to represent railway traffic, and to resolve any conflicts that occur, is the large computational time needed to obtain satisfactory results. Therefore, the purpose of this article is to study and design a discrete event-based model, characterized by the positioning of trains in block sections, that can represent the rail system, including the dynamic aspect, and a fixed block signaling system able to proactively detect and resolve potential conflicts that may occur within this system. This model can be developed and integrated as a specific module of a railway passenger-dedicated intelligent transportation systems (ITS) oriented to real-time traffic management in order to improve the service quality of a railway system's operations. The aim is to reduce the computational cost as much as possible and implement the proposed model in a railway network. A numerical investigation based on the Renfe Cercanís Madrid rail network (Spain) shows the high computational performance of the proposed approach in a real-life application context. [ABSTRACT FROM PUBLISHER]

An erasure correction strategy based on fountain coding is proposed for traffic with real-time requirements. A sliding window marks the range of non-expired data. Each new block entering the window is once sent as such, followed by probabilistically sending a repair packet. The repair packets are formed as a random combination of the blocks in the current window using a degree distribution as in LT coding. The performance of the method with a given channel loss probability is analyzed using a Markov chain model. The state space, however, has to be truncated for computational tractability. The truncation error is verified to be small enough by simulations. By using the analytical model the optimal degree distribution is found to be of single-degree type. The performance of the proposed scheme is compared with deterministic settings, in which repair packets are sent after fixed number of systematic packets. Further comparison is made against Raptor coding, and we note that using the presented strategy can result in better performance in some situations. [ABSTRACT FROM AUTHOR]

Due to multiple hops, mobility and time-varying channel, supporting delay sensitive real-time traffic in wireless local area network-based (WLAN) mesh networks is a challenging task. In particular for real-time traffic subject to medium access control (MAC) layer control overhead, such as preamble, carrier sense waiting time and the random backoff period, the performance of real-time flows will be degraded greatly. In order to support real-time traffic, an efficient adaptive packet scheduling (APS) scheme is proposed, which aims to improve the system performance by guaranteeing inter-class, intra-class service differentiation and adaptively adjusting the packet length. APS classifies incoming packets by the IEEE 802.11e access class and then queued into a suitable buffer queue. APS employs strict priority service discipline for resource allocation among different service classes to achieve inter-class fairness. By estimating the received signal to interference plus noise ratio (SINR) per bit and current link condition, APS is able to calculate the optimized packet length with bi-dimensional markov MAC model to improve system performance. To achieve the fairness of intra-class, APS also takes maximum tolerable packet delay, transmission requests, and average allocation transmission into consideration to allocate transmission opportunity to the corresponding traffic. Detailed simulation results and comparison with IEEE 802.11e enhanced distributed channel access (EDCA) scheme show that the proposed APS scheme is able to effectively provide inter-class and intra-class differentiate services and improve QoS for real-time traffic in terms of throughput, end-to-end delay, packet loss rate and fairness. [ABSTRACT FROM AUTHOR]

In the metropolitan region, most congestion or traffic jams are caused by the uneven distribution of traffic flow that creates bottleneck points where the traffic volume exceeds the road capacity. Additionally, unexpected incidents are the next most probable cause of these bottleneck regions. Moreover, most drivers are driving based on their empirical experience without awareness of real-time traffic situations. This unintelligent traffic behavior can make the congestion problem worse. Prediction based route guidance systems show great improvements in solving the inefficient diversion strategy problem by estimating future travel time when calculating accurate travel time is difficult. However, performances of machine learning based prediction models that are based on the historical data set degrade sharply during a congestion situation. This paper develops a new navigation system for reducing travel time of an individual driver and distributing the flow of urban traffic efficiently in order to reduce the occurrence of congestion. Compared with previous route guidance systems, the results reveal that our system, applying the advanced multi-lane prediction based real-time fastest path (AMPRFP) algorithm, can significantly reduce the travel time especially when drivers travel in a complex route environment and face frequent congestion problems. Unlike the previous system,1 it can be applied either for single lane or multi-lane urban traffic networks where the reason for congestion is significantly complex. We also demonstrate the advantages of this system and verify the results using real highway traffic data and a synthetic experiment. [ABSTRACT FROM AUTHOR]

This paper studies the problem of congestion control and scheduling in ad hoc wireless networks that have to support a mixture of best-effort and real-time traffic. Optimization and stochastic network theory have been successful in designing architectures for fair resource allocation to meet long-term throughput demands. However, to the best of our knowledge, strict packet delay deadlines were not considered in this framework previously. In this paper, we propose a model for incorporating the quality-of-service (QoS) requirements of packets with deadlines in the optimization framework. The solution to the problem results in a joint congestion control and scheduling algorithm that fairly allocates resources to meet the fairness objectives of both elastic and inelastic flows and per-packet delay requirements of inelastic flows. [ABSTRACT FROM PUBLISHER]

We investigate the problem of delay constrained maximal information collection for CSMA-based wireless sensor networks. We study how to allocate the maximal allowable transmission delay at each node, such that the amount of information collected at the sink is maximized and the total delay for the data aggregation is within the given bound. We formulate the problem by using dynamic programming and propose an optimal algorithm for the optimal assignment of transmission attempts. Based on the analysis of the optimal solution, we propose a distributed greedy algorithm. It is shown to have a similar performance as the optimal one. [ABSTRACT FROM AUTHOR]

In this paper, our recent work interval-based orderly scheduling strategy (IOSS) is extended by adding quality-of-service (QoS) provisioning. A revised medium access control (MAC) protocol is introduced, which supports priority-based QoS in wavelength division multiplexing (WDM) networks with star topology. The proposed interval-based prioritized orderly scheduling strategy (IPOSS) operates in a distributed manner, and has the capability of handling channel collision and destinations conflicts in order to provide a collision-free scheduling scheme. Each node in the network is equipped with a tunable transmitter and a fixed receiver, assuring that the scheme is scalable with respect to the number of nodes and channels. Also, each node may accept high- and low-priority packet arrivals. IPOSS favors high-priority packets, without regarding packets' length or packets' destination. Moreover, the proposed scheme differentiates the packet's schedule order by prioritizing the long-length over the short-length packets. It is found that the adopted access control scheme achieves a critically high throughput-delay performance for real-time traffic. Furthermore, IPOSS presents a little bit improved throughput performance than IOSS scheme, since it handles in a different way the case in which two or more requests demand the same amount of transmission time. The traffic involved in the simulation results follows Bernoulli and Poisson distribution, regarding the arrival of the requests. [ABSTRACT FROM PUBLISHER]

EDCA (Enhanced Distributed Channel Access), developed for providing service differentiation in wireless LANs, has become a MAC standard in IEEE 80.2.11e. However, unexpected packet delay due to collisions and retransmissions may occur when the wireless LAN becomes congested. To support the increasing demand of delay-sensitive multimedia applications, this paper presents a modified EDCA with dynamic contention control (DCC) for real-time traffic in multi-hop ad hoc networks. The proposed scheme not only reduces the average packet delay but also increases the percentage of packets meeting real-time constraints by two approaches: (1) dynamically adjusting the priority level of a traffic flow based on the estimated per-hop delay, and (2) generating a nonuniformly distributed backoff timer for retransmitted frames according to their individual end-to-end delay requirements. For the purpose of evaluation, we perform simulations on ns-2. We compare the performance of the proposed DCC scheme to EDCA and a previous work, AEDCF (adaptive EDCF). The simulation results demonstrate the effectiveness and superiority of our proposed DCC scheme. [ABSTRACT FROM AUTHOR]

Currently, there is a trend towards the implementation of industrial communication systems using wireless networks. However, keeping up with the timing constraints of real-time traffic in wireless environments is a hard task. The main reason is that real-time devices must share the same communication medium with timing unconstrained devices. The VTP-CSMA architecture has been proposed to deal with this problem. It considers an unified wireless system in one frequency band, where the communication bandwidth is shared by real-time and non-real-time communicating devices. The proposed architecture is based on a virtual token passing (VTP) procedure that circulates a virtual token among real-time devices. This virtual token is complemented by an underlying traffic separation mechanism that prioritizes the real-time traffic over the non-real-time traffic. This is one of the most innovative aspects of the proposed architecture, as most part of real-time communication approaches are not able to handle timing unconstrained traffic sharing the same communication medium. A ring management procedure for the VTP-CSMA architecture is also proposed, allowing real-time stations to adequately join/leave the virtual ring. [ABSTRACT FROM PUBLISHER]

Based on the requirements of beyond 3G mobile communication systems, the access network architecture and media access control technique for B3G systems are studied. The proposed novel access network architecture, which can reduce network complexity and improve system performance, is introduced. Centralized mini-slot packet reservation multiple access (CMPRMA), based on OFDMA, is proposed, which not only can inherit the advantages of MPRMA and support real-time traffic well, but can also supply the resource reservation scheme for data traffic and support transmission for data traffic efficiently [ABSTRACT FROM PUBLISHER]

As the wireless environment becomes ever more populated and complex, individual networks offering single or restricted services will become ever more uncompetitive. Common radio resource management algorithms and strategies are used in heterogeneous or converged wireless networks to integrate multiple physical radio interfaces to support different levels of data rates, mobility, and traffic. Admission control in a common RRM environment is used to select the most appropriate wireless access based on service type, user preference, and network load. This is used to provide a balance between real-time and non-real-time traffic across the available access technologies. The development of common radio resource management strategies requires a suitable platform of coupled access networks and is the motivation behind the development of a converged wireless test platform. The tightly coupled platform supports voice and data call setup and delivery over UMTS and WLAN [ABSTRACT FROM PUBLISHER]

Recent advances in signal processing techniques have enabled wireless networks to have multi-packet reception (MPR) capability at the physical layer, where it is possible to receive more than one packets when concurrent transmissions occur. In this paper, we propose a novel multi-reservation multiple access (MRMA) scheme for future wireless multimedia networks based on such an MPR channel model. MRMA can fully exploit the channel's MPR capacity while fulfilling the quality of service requirements of different multimedia traffic. Similar to many existing reservation protocols, MRMA guarantees (i) bounded access delay for real-time traffic by dropping packets not transmitted within the required delay bound, and (ii) satisfaction of the required packet loss ratio when applied in conjunction with a connection admission control method to limit the number of admitted traffic streams. MRMA further incorporates a priority control mechanism to differentiate guaranteed and best-effort services for real-time voice/video and connectionless data, respectively. Analytical and simulation results are presented to illustrate the effectiveness of MRMA in service differentiation and increasing the number of admitted traffic streams [ABSTRACT FROM PUBLISHER]

This study proposes a new multiple access control protocol named distributed synchronous reservation multiple access control protocol, in which the hidden and exposed terminal problems are solved, and the quality of service (QoS) requirements for real-time traffic are guaranteed. The protocol is founded on time division multiplex address and a different type of traffic is assigned to different priority, according to which a node should compete for and reserve the free slots in a different method. Moreover, there is a reservation acknowledgement process before data transmit in each reserved slot, so that the intruded terminal problem is solved. The throughput and average packets drop probability of this protocol are analyzed and simulated in a fully connected network, the results of which indicate that this protocol is efficient enough to support the real-time traffic, and it is more suitable to MANETs. [ABSTRACT FROM AUTHOR]

The IEEE 802.16 standard defines three types of scheduling services for supporting real-time traffic, unsolicited grant service (UGS), real-time polling service (rtPS), and extended real-time polling service (ertPS). In the UGS service, the base station (BS) offers a fixed amount of bandwidth to a subscriber station (SS) periodically, and the SS does not have to make any explicit bandwidth requests. The bandwidth allocation in the rtPS service is updated periodically in the way that the BS periodically polls the SS, which makes a bandwidth request at the specified uplink time slots and receives a bandwidth grant in the following downlink subframe. In the ertPS service, the BS keeps offering the same amount of bandwidth to the SS unless explicitly requested by the SS. The SS makes a bandwidth request only if its required transmission rate changes. In this article we study the performance of voice packet transmissions and BS resource utilization using the three types of scheduling services in IEEE 802.16-based backhaul networks, where each SS forwards packets for a number of voice connections. Our results demonstrate that while the UGS service achieves the best latency performance, the rtPS service can more efficiently utilize the BS resource and flexibly trade-off between packet transmission performance and BS resource allocation efficiency; and appropriately choosing the MAC frame size is important in both the rtPS and ertPS services to reduce packet transmission delay and loss rate [ABSTRACT FROM PUBLISHER]

Abstract: This study aims at ‘predicting’ the occurrence of lane-change related freeway crashes using the traffic surveillance data collected from a pair of dual loop detectors. The approach adopted here involves developing classification models using the historical crash data and corresponding information on real-time traffic parameters obtained from loop detectors. The historical crash and loop detector data to calibrate the neural network models (corresponding to crash and non-crash cases to set up a binary classification problem) were collected from the Interstate-4 corridor in Orlando (FL) metropolitan area. Through a careful examination of crash data, it was concluded that all sideswipe collisions and the angle crashes that occur on the inner lanes (left most and center lanes) of the freeway may be attributed to lane-changing maneuvers. These crashes are referred to as lane-change related crashes in this study. The factors explored as independent variables include the parameters formulated to capture the overall measure of lane-changing and between-lane variations of speed, volume and occupancy at the station located upstream of crash locations. Classification tree based variable selection procedure showed that average speeds upstream and downstream of crash location, difference in occupancy on adjacent lanes and standard deviation of volume and speed downstream of the crash location were found to be significantly associated with the binary variable (crash versus non-crash). The classification models based on data mining approach achieved satisfactory classification accuracy over the validation dataset. The results indicate that these models may be applied for identifying real-time traffic conditions prone to lane-change related crashes. [Copyright &y& Elsevier]

In the emerging IEEE 802.11e MAC protocol, the enhanced distributed channel access (EDCA) is proposed to support prioritized QoS; however, it cannot guarantee strict QoS required by real-time services such as voice and video without proper network control mechanisms. To overcome this deficiency, we first build an analytical model to derive an average delay estimate for the traffic of different priorities in the unsaturated 802.11e WLAN, showing that the QoS requirements of the real-time traffic can be satisfied if the input traffic is properly regulated. Then, we propose two effective call admission control schemes and a rate control scheme that relies on the average delay estimates and the channel busyness ratio, an index that can accurately represent the network status. The key idea is, when accepting a new real-time flow, the admission control algorithm considers its effect on the channel utilization and the delay experienced by existing real-time flows, ensuring that the channel is not overloaded and the delay requirements are not violated. At the same time, the rate control algorithm allows the best effort traffic to fully use the residual bandwidth left by the real-time traffic, thereby achieving high channel utilization [ABSTRACT FROM PUBLISHER]

Recently, polling has been included as a resource sharing mechanism in the medium access control (MAC) protocol of several communication systems, such as the IEEE 802.11 wireless local area network, primarily to support real-time traffic. Furthermore, to allow these communication systems to support multimedia traffic, the polling scheme often coexists with other MAC schemes such as random access. Motivated by these systems, we develop a model for a polling system with vacations, where the vacations represent the time periods in which the resource sharing mechanism used is a non-polling mode. The real-time traffic served by the polling mode in our study is telephony. We use an on-off Markov modulated fluid (MMF) model to characterize telephony sources. Our analytical study and a counterpart validating simulation study show the following. Since voice codec rates are much smaller than link transmission rates, the queueing delay that arises from waiting for a poll dominates the total delay experienced by a voice packet. To keep delays low, the number of telephone calls that can be admitted must be chosen carefully according to delay tolerance, loss tolerance, codec rates, protocol overheads and the amount of bandwidth allocated to the polling mode. The effect of statistical multiplexing gain obtained by exploiting the on-off characteristics of telephony traffic is more noticeable when the impact of polling overhead is small [ABSTRACT FROM PUBLISHER]

AM and FM radio has long been the primary source of mobile entertainment and information. Radio's conversion to digital broadcasting through deployment of the FCC approved HD radio technology with drive new and innovative telematics services. In the digital world, the radio will be transformed to a centralized source of expanded entertainment and information services along with enhanced in-vehicle systems. At a fraction of the cost of other wireless services, HD radio technology will provide platform-specific customer retention information, real-time traffic for navigation system overlay, PVR-like functionality, on-demand audio, mobile commerce and much more, dramatically upgrading automobile manufacturers' ability to reach and profit from radio's 200 million listeners per day [ABSTRACT FROM PUBLISHER]

In multi-service wireless networks, asymmetric bandwidth allocation has been proposed to satisfy the requirements of asymmetric traffic load introduced by some data applications. However, it is difficult to promptly adjust bandwidth allocation on uplink and downlink according to the dynamics of traffic load. Inappropriate call admission control (CAC) policies in this environment may admit superfluous real-time (RT) calls or non real-time (NRT) calls and thus lead to low bandwidth utilization. In this paper, we propose and evaluate two new CAC schemes to address the problems caused by the mismatch of bandwidth allocation and traffic changing in multi-service wireless networks with bandwidth asymmetry between uplink and downlink. By determining the admissible regions for the RT calls and the NRT calls, the proposed schemes prevent the calls of a specific class from overusing the bandwidth resources. Mathematical analysis and simulation experiments are employed to study and compare the performance of the proposed schemes and the existing schemes. The numerical results show that the proposed schemes can achieve better performances in terms of call blocking probability and bandwidth utilization compared to some existing schemes, even those performing well in bandwidth asymmetry wireless networks. [ABSTRACT FROM AUTHOR]

Abstract: An ad hoc wireless network (AWN) is a collection of mobile hosts forming a temporary network on the fly, without using any fixed infrastructure. Characteristics of AWNs such as lack of central coordination, mobility of hosts, dynamically varying network topology, and limited availability of resources make QoS provisioning very challenging in such networks. In this paper, we describe the issues and challenges in providing QoS for AWNs and review some of the QoS solutions proposed. We first provide a layer-wise classification of the existing QoS solutions, and then discuss each of these solutions. [Copyright &y& Elsevier]

While backbone networks have experienced substantial changes in the last decade, access networks have not changed much. Recently, passive optical networks (PONs) seem to be ready for commercial deployment as access networks, due to the maturity of a number of enabling technologies. Among the PON technologies, Ethernet PON (EPON) standardized by the IEEE 802.3ah Ethernet in the First Mile (EFM) Task Force is the most attractive one because of its high speed, low cost, familiarity, interoperability, and low overhead. In this paper, we consider the issue of upstream channel sharing in the EPONs. We propose a novel multiple-access control scheme to provide bandwidth-guaranteed service for high-demand customers, while providing best effort service to low-demand customers according to the service level agreement (SLA). The analytical and simulation results prove that the proposed scheme performs best in what it is designed to do compared to another well-known scheme that has not considered providing differentiated services. With business customers preferring premium services with guaranteed bandwidth and residential users preferring low-cost best effort services, our scheme could benefit both groups of subscribers, as well as the operators. [ABSTRACT FROM PUBLISHER]

One of the major challenges in the design of future-generation high-speed networks is the provision of quality-of-service (QoS) to real-time traffic. In this paper, we propose a novel scheduling scheme, namely differentiated dropping scheduling (DDS), which is designed to handle real-time traffic in passive star-coupled wavelength division multiplexing (WDM) optical networks. By taking channel and destination availability into account, DDS can dramatically improve network performance in terms of message loss rate. Moreover, this scheme has the capacity of preventing channel collision and destination conflict. In order to evaluate the proposed DDS algorithm, extensive discrete-event simulations and mathematical performance comparison are conducted by comparing its performance with Moore and Hodgson's algorithm and the earliest-due-date (EDD) algorithm. The results show that DDS can achieve the best performance among the three algorithms. [ABSTRACT FROM PUBLISHER]

Characteristics of Mobile Ad hoc Networks such as shared broadcast channel, bandwidth and battery power limitations, highly dynamic topology, and location dependent errors, make provisioning of quality of service (QoS) in such networks very difficult. The Medium Access Control (MAC) layer plays a very important role as far as QoS is concerned. The MAC layer is responsible for selecting the next packet to be transmitted and the timing of its transmission. We have proposed a new MAC layer protocol that includes a laxity-based priority scheduling scheme and an associated back-off scheme, for supporting time-sensitive traffic. In the proposed scheduling scheme, we select the next packet to be transmitted, based on its priority value which takes into consideration the uniform laxity budget of the packet, the current packet delivery ratio of the flow to which the packet belongs, and the packet delivery ratio desired by the user. The back-off mechanism devised by us grants a node access to the channel, based on the rank of its highest priority packet in comparison to other such packets queued at nodes in the neighborhood of the current node. We have studied the performance of our protocol that combines a packet scheduling scheme and a channel access scheme through simulation experiments, and the simulation results show that our protocol exhibits a significant improvement in packet delivery ratio under bounded end-to-end delay requirements, compared to the existing 802.11 DCF and the Distributed Priority Scheduling scheme proposed recently in [ACM Wireless Networks Journal 8 (5) (2002) 455–466; Proceedings of ACM MOBICOM ''01, July 2001, pp. 200–209]. [Copyright &y& Elsevier]

MobiWeb is a proxy-based network architecture designed to enhance the performance of adaptive real-time streams over 3G wireless links. MobiWeb includes a priority scheme that preserves media smoothness despite short-term fluctuations of the link, as well as an adaptation mechanism applied to streams responding to long-term link changes and handoffs. A set of timers permits the rapid exploration of the link, increasing the utilization of resources when they are abundant, while forcing enhanced streams to back off to more moderate resource usage when resources are scarce. MobiWeb uses admission control for real-time traffic in order to provide a base level of quality, while remaining transparent to unaware best-effort traffic. This article validates the enhanced performance of adaptive real-time streams when MobiWeb is used through the evaluation of several scenarios simulated on NS-the Network Simulator tool [ABSTRACT FROM PUBLISHER]

A new ATM AAL2 to support low-bit-rate and delay-sensitive applications in an ATM environment has been approved by the ITU-T and ATM Forum. AAL2 allows multiplexing of voice/data packets (minicells) from different users in a single ATM connection, thus improving the utilization of the network resources. This article identifies issues related to AAL2 such as the application environment, the gain obtained by multiplexing voice packets from different users on a single ATM connection, buffer requirements, and cell assembly delay. To analyze these issues, we modeled and simulated ATM trunking between a base transceiver station and the mobile switching center. The performance simulations show that the multiplexing gain obtained by AAL2 is significant, and the output buffer requirement at the ATM node is small. In addition, the results show that a timeout mechanism is needed to limit the ATM cell assembly delay in order to meet the delay requirements of real-time traffic such as voice [ABSTRACT FROM PUBLISHER]

This paper presents the architecture and performance of a new shared media ATM LAN protocol known as pipelined-access local ATM network (PLAN) which has high performance and operates in native ATM mode for efficient and simpler interworking with backbone ATM networks. PLAN is a reservation type bidirectional broadcast bus system protocol in which overhead due to bus propagation delay is limited to a total of twice the end-to-end propagation delay per transmission frame irrespective of the number of stations on the bus. It is designed to (i) support multimedia traffic, (ii) dynamically allocate bandwidth to various types of traffic and provide fair media access to stations, (iii) provide throughput close to 100%, (iv) operate at high bit rates and (v) use packets of ATM cell size and format for interoperability with wide-area ATM networks. In this paper we present the architecture, analysis and performance of the proposed PLAN protocol under various multimedia traffic conditions. The results show that its throughput approaches 100%, is independent of traffic composition and increases slowly with bit rate. It decreases only slightly as the bus length and/or number of stations increase. The proposed PLAN protocol is seen to be well suited for high-speed multimedia ATM LAN and MAN applications in the ATM networking environment. © 1998 John Wiley & Sons, Ltd. [ABSTRACT FROM AUTHOR]