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In the modern data networks, the demand is increasing for interruption free, high speed, and best quality of services to perform successful communications. This paper emphasizes on the high priority (HP) traffic like video conferencing, video streaming, telemedicine, teleconferencing, and interactive gaming for the real time communications. To achieve the best quality data communication for the HP traffic, in this paper, we propose a new dynamic bandwidth allocation (DBA) algorithm called dynamic hybrid slot-size bandwidth allocation (DHSSBA) algorithm, which is the modified version of existing hybrid slot-size/rate (HSSR) algorithm. The performance of the proposed scheme has been analyzed by using numerical simulations in terms of the end-to-end packet delay, absolute delay variation, jitter, and throughput for both the HP and best effort (BE) traffic. We have also compared the simulation results of the proposed scheme with those of the existing modified HSSR (MHSSR), delay variation guaranteed protocol (DVGP), and HSSR schemes. From the comparison of the simulation results, at the highest offered load of 1.0, it can be seen that for the HP traffic the proposed scheme provides 60%, 50%, and 20% less average end-to-end delay than the existing HSSR, DVGP, and MHSSR schemes, respectively. At the highest offered load of 1.0, the proposed scheme also provides about 35% higher throughput than both the HSSR and DVGP schemes with 45% and 35% less jitter variation than the HSSR and DVGP schemes, respectively.

Autonomous vehicles are in full development and vehicles classification is a fundamental part of this new technology in this years. The AVs is very is important solution in smart cities. The electric vehicle EV is one of the solutions recommended by the vehicle manufacturers and research organizations to reduce noise pollution, fuel consumption, real time and time execution for tasks in networking EV. The transport and mobility sector has for more than a decade seen a fundamental change in its organization due to a double technological revolution and use a new forms of mobility. This motion has profound consequences on the relationship that citizens maintain with mobility. It also proposes a physical platform able to perform a form of platooning using Artificial Intelligence (AI) with scheduler to create platoons with miniature vehicle. Platform called Autonomous Learning Intelligent Vehicles is the association of multiple physical cars coupled with an infrastructure, which handles the data of every single car and makes decisions based on the augmented environmental. In this solution, we present tracking and detection of EVs in parking or in station. So, we change communications EVs, giving priority to charging demands over other types of sms lower priority messages. Finally, we propose an efficient admission control mechanism to manage EVs traffic and to provide quality of Service to charging demand messages in terms of strict delay to avoid both a long latency of EV users and a network overload in high offered load conditions.

The article proposes a new method of determining traffic characteristics of multiservice overflow systems that carry adaptive traffic. When the total offered load in primary resources exceeds a certain value, this type of traffic is admitted for service with lower bitrate. A particular attention is given in the article to a method for a determination of the parameters of traffic that overflows to secondary resources as well as to the way adaptive traffic is serviced. The method takes into consideration three possible types of traffic: Erlang, Engset, and Pascal traffic. It is based on a generalization of Hayward’s concept and its application to model systems with adaptive traffic with threshold compression. The method can be used for optimal dimensioning of logical networks (slices) in modern mobile systems due to possibility of analytical determination of grade of service parameters (blocking probability, carried traffic, and network load). To verify the accuracy of the proposed model the results of analytical calculations, obtained on the basis of the proposed model, are then compared with the results of simulation experiments for a number of selected structures of overflow systems that service adaptive traffic. The results of the study demonstrate high accuracy of the proposed theoretical model.

Voice over Internet Protocol (VoIP) uses packet switching to transmit voice, video, data and chat services. VoIP applications depend upon various speech codecs which vary in speech quality, bandwidth and computational requirements. The key challenges for improving the VoIP Quality of Service (QoS) and Quality of Experience (QoE) is to use best codec by considering network traffic and selecting the optimal hardware. Hardware Calibration is a mechanism to aid service providers by selecting a suitable hardware for Call Manager to transmit different real-time based applications. The paper focuses on proposing a computational model which suggests the most suitable hardware for a specific load handling requirement. The selection of processor is done by taking into consideration the effect of different codecs when it takes calls per second as input and gives output as whichever hardware is recommended for Call Manager to handle different types of applications like Peer-To-Peer (P2P) application, Back-To-Back (BTB) application, enabling voice logging, etc. The emulations demonstrate that the proposed model is suitable for selection of hardware which provides better QoS and QoE for the transmission VoIP based real-time applications. Further, a multi-criteria decision-making method based on Analytic Hierarchy Process (AHP) is also proposed to decide which hardware can be chosen as Call Manager. The results obtained from AHP are used to validate the outcome of the proposed computational model. The computational model is applicable only when the tests are run on particular systems and it suggests the best hardware among them for the particular offered load. However, to suggest the best configuration corresponding to the input load from the possible configurations available in the market, benchmarking is performed. A common scale model for hardware benchmark estimation particularly has been proposed for different scenarios.

Sensor sources submit updates to a monitor through an unslotted, uncoordinated, unreliable multiple access collision channel. The channel is unreliable; a collision-free transmission is received successfully at the monitor with some transmission success probability. For an infinite-user model in which the sensors collectively transmit updates as a Poisson process and each update has an independent exponential transmission time, a stochastic hybrid system (SHS) approach is used to derive the average age of information (AoI) as a function of the offered load and the transmission success probability. The analysis is then extended to evaluate the individual age of a selected source. When the number of sources and update transmission rate grow large in fixed proportion, the limiting asymptotic individual age is shown to provide an accurate individual age approximation for a small number of sources., This has been submitted to ISIT 2020

Understanding delay performance is essential for successful development and deployment of real-time networked sensing applications. In this work, we characterize the end-to-end delay on a multi-hop Wireless Sensor Network (WSN) with an analytical method based on Stochastic Network Calculus (SNC) with Moment Generating Functions (MGF). The particularity of the sensing scenario is that all nodes generate and cooperatively forward their traffic through the network to a main location, known as the sink. We tackled the problem by modeling the service process of fading wireless channels from a high-layer perspective with Finite-State Markov Chains (FSMC). Numerical performance bounds are provided for an example WSN with IEEE 802.11g ad-hoc links in which the effects of delay bound violation probability, per-node offered load, data rate and fading speed are quantified for different network sizes. Finally, the presented analysis is validated through a comparison between analytical and numerical simulation results.

Recently, high port-count optical packet switches, based on the Arrayed Waveguide Grating (AWG) device, have been proposed for providing the very high bandwidth and connectivity degree that will be required in future datacentre interconnection networks. A practical, cost-effective realisation of such a switch requires that over-subscription of the AWG cross-connect output ports be allowed, with the resulting packet contention being resolved using a packet buffering facility in the switch. In this paper, we propose an efficient AWG switch architecture of this type that uses a recirculating optical packet buffering scheme and we develop analytic modelling methods to estimate the switch packet blocking probability and packet buffering delay. The analytic model resolves internal traffic flows in the switch as Markov-modulated Poisson Processes (MMPP) and also allows modelling of offered load to the switch as a MMPP process. Taking this model of the switch, we first show its accuracy compared to discrete event simulations and then use it to develop a heuristic optimisation algorithm to dimension the switch hardware components, with the objective of minimising the operating cost by optimally dimensioning the main power consuming components, subject to constraints on the maximum contention probability and average and maximum propagation delays. The resulting optimised switch design demonstrates the proposed architecture to be a promising candidate for future high-port count datacentre switches.

Accurate prediction of traffic conditions on orthogonal frequency division multiple access passive optical networks is important because of its vital role in network resource management and efficient bandwidth allocation. Given the dynamic and stochastic nature of network traffic, our proposed algorithm conducts a probabilistic approach by using the hidden Markov model (HMM). The HMM defines traffic states with two parameters: the mean and contrast of the bandwidth request observations. Simulation results demonstrate the performance comparison between with and without the prediction method in terms of throughput and end-to-end delay. As a result, the throughput improves 15% and the saturation offered load of the delay for the prediction and non-prediction is 0.8 and 0.7, respectively.

In this paper, we propose two task admission control algorithms that utilize random task filtering: a lightweight algorithm based on long-term estimates of average utilization and offered load, and a more complex algorithm based on instantaneous utilization. Detailed performance evaluation confirms that both algorithms are able to ensure that the system remains in the stable operating region. We have also found that more aggressive filtering tends to decrease task blocking rate and delay. Copyright © 2016 John Wiley & Sons, Ltd.

Unlike terrestrial networks that mainly rely on radio waves for communications, underwater networks utilize acoustic waves, which have comparatively lower loss and longer range in underwater environments. However, acoustic waves incurs long propagation delays that typically lead to low throughput and higher energy cost of transmission than reception. Thus, collision and retransmission should be reduced in order to reduce energy cost and improve throughput. Receiver-based protocols, like receiver-initiated packet train and cluster-based on-demand time sharing, can significantly reduce collision and retransmission. But they are either time-consuming or energy consuming because nodes are controlled to turn into receiver mode by control packets or a timer regardless of load. In this paper, we propose an underwater practical MAC protocol (UPMAC), whose objective is to be adaptive to the network load conditions by providing two modes (high and low load modes) and switching between them based on different offered load. Turn-around time overhead is reduced and it is less vulnerable to control packet corruption by using the technique of piggyback. UPMAC uses receiver-based approach in high load mode, leading to a low data collision rate in both one-hop and multihop situation. Extensive simulations show that UPMAC can achieve better performance in both general and Sea Swarm topologies.

At the base station of a wireless broadband network, connection requests for various services viz., voice, browsing, streaming video, online gaming etc., arrive at random points of time and depart at random points of time. The arrival rates of connection requests for various services change with time of the day, resulting in non-stationary aggregate traffic load that is mathematically complex to model. We propose a novel approach, by using two techniques of stochastic point processes viz., Product Density and Random Point Processes to model time dependent problems in wireless broadband networks. We classify the problems faced by a wireless broadband service provider into two categories---counting measures and aggregate measures. By counting measures, we mean expected number and fluctuation in (a) number of calls of each traffic type in service at any time of the day (b) number of calls that have completed service in a time interval (c) number of calls that have exceeded a given delay threshold and (d) offered load. We define Product Densities of first order to evaluate the expected value and Product Densities of second order to evaluate the fluctuation of these measures for non-homogeneous Poisson arrival rate and exponential servicing rate. Each of the connection requests arriving at random instants of time triggers a random process of traffic, for a random duration. The cumulative built-up of the triggered random process is non-Markovian and hence, it is mathematically complex to estimate its statistical characteristics. We categorize as aggregate measures, the expected value and fluctuation in (a) aggregate traffic load at the base station from all types of traffic and (b) aggregate resources (or bandwidth) required to service this traffic load. The cumulative built-up of these measures fits in with Random Point Process, a special class of stochastic point processes. We estimate the expected value and fluctuation of aggregate measures for time dependent arrival rates, using Random Point Process and Product Density.

The efficiency of service of Underwater Acoustic Networks is restricted by the acoustic channel characteristic, such as the low bandwidth , high varying multipath and elevated noise levels. In addition, the lifetime of Underwater Acoustic Networks is influenced greatly by the limited battery power. This paper designs an Underwater Acoustic Networks and poses the data link layer protocols including the logical link control protocol and the media access control protocol. The simulation results have demonstrated that the protocols are of certain applicability in underwater acoustic channel environment that maximize throughput and reliability while minimizing end-to-end delay. In the event of low offered load, the protocol can satisfy the requirment of Underwater Acoustic Networks.

HTTP progressive video downloading is becoming increasingly relevant in wireless and mobile networks. Therefore, improving the quality of experience (QoE) for this service is important for customer satisfaction. In this letter we propose a QoE-aware scheduling solution for the HTTP progressive video downloading in OFDMA systems. Whereas most scheduling strategies consider QoS metrics such as delay, jitter or throughput, our proposal takes into account a metric that directly affects the end-users experience. In particular, we propose to use an estimation of the amount of video data stored in the player buffer as the main criterion for resource assignment. The ultimate goal of using this criterion is the avoidance of playback interruptions. In particular, we have evaluated this scheduler in an LTE network simulator for the YouTube service, which is the most visited HTTP video streaming site. Our results show a significant improvement compared to the Multicarrier Proportional Fair scheduler, e.g. the number of videos with pauses decrease from 16.1% to 0% for 12 Mbps of cell offered load.

In Mobile Wireless Sensor Networks (MWSN), during routing, there may be an increase in congestion levels. It is required to automatically alter the traffic rate accordingly. In this paper, we propose to design an intra and inter cluster congestion control technique for MWSN. In intra-cluster congestion control, the Cluster Head (CH) proactively monitors the congestion within the cluster each time based on parameters such as traffic intensity information, buffer occupancy and number of contenders. If the congestion level is more than the threshold, it then modifies the traffic rate based on the offered load and the number of contenders. In inter-cluster congestion control, the gateway or forwarder nodes choose the best CH based on the load. By simulation results, we show that the proposed technique minimizes the congestion and the packet drops through effective regulation of the network load.

To perform an Optical Burst Switching (OBS) network, a lot of challenges are faced that have to be solved. The edge node has to manage burst offset time and burst assembly technique. The core node has to reserve resources to send the data burst, schedule the control packet and data burst and contention resolution strategy. In this paper, we discuss the burst assembly mechanisms and their disadvantages which we will overcome using a new and simple algorithm for data burst generation which relies on changing some characteristics in queuing model in OBS network by taking into consideration the presences of Quality of Service (QoS) for multiple data priorities. The OBS network simulation using our algorithm shows that it adaptively changes the data burst size according to the offered load. So, it can offer high data burst utilization, limits the maximum end to end delay and reduces burst drop rate for high priority packets.

In this paper, we propose a scheme for partially dynamic lane control for energy saving in multilane-based high-speed Ethernet. In this scheme, among the given transmission lanes, at least one lane is always operating, and the remaining lanes are dynamically activated to alleviate the network performance in terms of queuing delay and packet loss in the range of acceptance. The number of active lanes is determined by the decision algorithm based on the information regarding traffic and queue status. The reconciliation sublayer adjusts the transmission lane with the updated number of lanes received from the algorithm, which guarantees no processing delay in the media access control layer, no overhead, and minimal delay of the exchanging control frames. The proposed scheme is simulated in terms of queuing delay, packet loss rate, lane changes, and energy saving using an OPNET simulator. Our results indicate that energy savings of around 55% (or, when the offered load is less than 0.25, a significant additional savings of up to 75%) can be obtained with a queuing delay of less than 1 ms, a packet loss of less than 10 –4 , and a control packet exchange time of less than 0.5 μs in random traffic.

Fibre delay lines (FDLs) can substantially reduce the burst loss in Optical Burst Switching (OBS) networks and share-per-node FDL configurations can provide a more cost-efficient solution compared to architectures where delay lines are shared per port. Nevertheless, mathematical performance analysis of this configuration is more difficult due to traffic correlations arising from the shared resource. In this paper, an approximate two-moment traffic model is developed for quantifying end-to-end burst blocking probability in networks of OBS switches with share-per-node FDLs. The two-moment approach can improve model accuracy over more usual Poisson network analysis methods and additionally allows the characteristics of offered load to be taken into account. The accuracy of the proposed method is shown to be favourable, when compared to discrete-event simulations of an OBS network.

In multi-hop ad hoc networks, sources may pump more traffic into the networks than that can be supported, resulting in high end-to-end delay and packet-loss rate. Controlling the offered load at the sources can eliminate this problem. To conduct traffic control, the throughput of the network is necessary. In addition, the authors propose a concept of bandwidth mapping to analyse how much traffic in application can be injected to make the network supporting the maximum throughput. A multi-dimensional Markov model is built to analyse the performance of IEEE 802.11 DCF in unsaturated condition, in which M/G/1/K queuing model is used to analyse the service condition of the packet queue inside the node. The effects of carrier sensing property, hidden-node problem and signal capture property are considered together to analyse the contention experienced by the nodes in the network. Furthermore, the bandwidth mapping situation of voice and video traffic is analysed in string topology of a multi-hop ad hoc network. This analysis can provide a theoretical guide for the traffic admission control.

Recently, as the interest in Green IT is exponentially increased, EPON with sleep mode has been studied to reduce energy consumption in access network. In oder to guarantee cyclic sleep for ONU(Optical Network Units), EPON with sleep mode transmits upstream and downstream data at the same time. However, since conventional algorithms for sleep mode in EPON allocate bandwidth to each ONU according to upstream bandwidth request, the QoS of downstream data is not guaranteed when the offered load of OLT is larger than that of ONU. In this paper, we propose a bandwidth allocation algorithm for improving QoS in EPON with sleep mode. The proposed algorithm compares its size with an upstream request of ONU when a downstream buffer in the OLT exceeds a QoS threshold. And then it allocates selectively a bandwidth that satisfies the required QoS between the bandwidth request of ONU and OLT. Therefore, the proposed algorithm can save energy through cyclic sleep of ONUs while guaranteeing the QoS of up/downstream data. In order to evaluate the proposed algorithm, we perform simulation in terms of total sleep time of ONUs, queueing delay between OLT and ONU, and the utilization of allocated bandwidth at OLT through OPNET.

IEEE 802.15.6 standard technology is proposed for low-power wireless communication in, on and around body, where vital signs such as pulse, blood pressure, ECG, and EEG signals are transmitted as a type of data packet. Especially, these vital signs should be delivered in real time, so that the latency from slave node to hub node can be one of the pivotal performance requirements. However, in the case of IEEE 802.15.6 technology data retransmission caused by transmission failure can be done in the next superframe. In order to overcome this limitation, we propose an adaptive polling algorithm for IEEE 802.15.6 technology. The proposing algorithm makes the hub to look for an appropriate time period in order to make data retransmission within the superframe. Through the performance evaluation, the proposing algorithm achieves a 61% and a 73% latency reduction compared to those of IEEE 802.15.6 technology in the environment of 70% traffic offered load with 10ms and 100ms superframe period. In addition, the proposing algorithm prevents bursty traffic transmission condition caused by mixing retransmission traffic with the traffic reserved for transmission. Through the proposing adaptive polling algorithm, it will be possible to transmit time-sensitive vital signs without severe traffic delay.

We consider a first-come first-served multiserver queue in the Quality- and Efficiency-Driven (QED) regime. In this regime, which was first formalized by Halfin and Whitt, the number of servers N is not small, servers' utilization is [Formula: see text] (Efficiency-Driven) while waiting time is [Formula: see text] (Quality-Driven). This is equivalent to having the number of servers N being approximately equal to [Formula: see text], where R is the offered load and β is a positive constant. For the G/DK/N queue in the QED regime, we analyze the virtual waiting time VN(t), as N increases indefinitely. Assuming that the service-time distribution has a finite support (hence the DK in G/DK/N), it is shown that, in the limit, the scaled virtual waiting time [Formula: see text] is representable as a supremum over a random weighted tree (S denotes a service time). Informally, it is then argued that, for large N, [Formula: see text]here [Formula: see text] is the averaging of [Formula: see text] over S, and the process [Formula: see text] is zero-mean Gaussian that summarizes all relevant information about arrivals and service times ([Formula: see text] arises as a limit of an infinite-server (G/DK/∞) process, appropriately scaled). The results are obtained by using both combinatorial and probabilistic arguments. Possible applications of our approximations include fast simulation of queues and estimation/prediction of customer waiting times in the QED regime.

In the emerging smart grid technologies, resource utilization is diversified and responsive demand became an important tool in availing various benefits. The informed load curtailment is a popular means of demand response aspect with customers being facilitated by financial incentives for offered load reduction. Therefore, augmentation of such responsive loads with unit commitment process can be effective in resource scheduling of smart grid. This paper attempts to develop a binary fireworks algorithm application solve unit commitment problem along with demand response and wind energy. While the wind energy is considered as non schedulable, load curtailment is modelled as schedulable through demand response aggregator. Further, system and thermal unit constraints applicable for wind energy and DR are modified. The proposed methodology is tested on system with coal fired thermal generation, demand response aggregator and wind energy. Various possible scheduling scenarios are laid out for examination. Simulation results are presented and discussed in detail. Results shows the effectiveness of demand response in reducing the operational cost of system operator. Also, the demand response showed betterment when scheduled along with wind energy. The reduction in cost from from traditional thermal unit scheduling is observed to be 6.5% which is best among all the possible scenarios.

The Random early detection (RED) active queue management (AQM) scheme uses the average queue size to calculate the dropping probability in terms of minimum and maximum thresholds. The effect of heavy load enhances the frequency of crossing the maximum threshold value resulting in frequent dropping of the packets. An adaptive queue management with random dropping (AQMRD) algorithm is proposed which incorporates information not just about the average queue size but also the rate of change of the same. Introducing an adaptively changing threshold level that falls in between lower and upper thresholds, our algorithm demonstrates that these additional features significantly improve the system performance in terms of throughput, average queue size, utilization and queuing delay in relation to the existing AQM algorithms., Comment: 17 pages, 19 figures, and submitted to Telecommunication Systems, Springer

In this paper, we propose the optimal offset-time decision (OOD) algorithm which can effectively reduce the pre-transmission delay compared to the perfect isolation model, and can also be extended to general n priority classes while the target loss probability of each class is guaranteed for the variable offered load. In order to drive the OOD algorithm, we first analyze the loss probability of each priority class through class aggregation and iteration method; the analytic results obtained through the proposed algorithm are then validated with results garnered from extensive simulation tests.

In this paper we propose a task admission control scheme in a cloud system. We develop two task admission control algorithms which are based on a random task filtering policy. First, we present a lightweight task admission control algorithm which is appropriate for the cloud systems with smooth changes of task arrival rate. This algorithm is based on long-term estimation of average utilization and offered load. Second, we introduce a task admission algorithm which is efficient for highly dynamic systems and is based on instantaneous utilization. The performance of the proposed schemes is evaluated against varying intensities of offered load. Performance evaluation of the proposed schemes confirms that both of them are able to ensure that the system is kept in the stable operating region.

Because of the high attenuation, sound is mainly used as the communication medium in underwater sensor networks instead of light or microwave. MAC protocols designed for radio networks aren't suitable for acoustic channels which feature long propagation delay and limited bandwidth. In this paper, we propose a location-based MAC protocol, called LT-MAC. LT-MAC dynamically determines each node's sending time according to the variable offered load, and assigns the time slot to each node based on the minimum waiting time rule. In addition, the protocol combines sending and acknowledgement to overcome low throughput and long end-to-end delay problems. By theoretical analysis and simulation, we show that LT-MAC is a comparatively good solution for single-hop and small-scale underwater sensor networks.

The percentages of blocking and forced termination rates as parameters representing quality of services (QoS) requirements are presented. The relation between the connection statistics of mobile users in a cell and the handoff number and new call number in next duration in each cell is explored. Based on the relation, statistic reservation tactics are raised. The amount of bandwidth for new calls and handoffs of each cell in next period is determined by using the strategy. Using this method can guarantee the communication system suits mobile connection request dynamic. The QoS parameters: forced termination rate and blocking rate can be maintained steadily though they may change with the offered load. Some numerical experiments demonstrate this is a practical method with affordable overhead.

A novel scheme, namely united stabilizing scheme for edge delay, is introduced in optical burst switched networks. In the scheme, the limits of burst length and assembly time are both set according to certain qualifications. For executing the scheme, the conception for unit input bit rate is introduced to improve universality, and the assembly algorithm with a buffer safety space under the self-similar traffic model at each ingress edge router is proposed. Then, the components of burst and packet delay are concluded, and the equations that limits of burst length and assembly time should satisfy to stabilize the burst edge delay under different buffer offered loads are educed. The simulation results show that united stabilizing scheme stabilizes both burst and packet edge delay to a great extent when buffer offered load changes from 0.1 to 1, and the edge delay of burst and packet are near the limit values under larger offered load, respectively.

We present a cost model for splitting Internet dial-up traffic (which varies by time-of-day) between two large modem banks. One of the modem banks charges by the hour, the other charges for the peak number-in-system during the day. To study if the possible savings are enough to make the effort worthwhile, we formulate a clairvoyant ("perfect information") Integer Program that is equivalent to a network flow problem. This leads us to use a ceiling policy. In the stochastic control case, we use a Modified Offered Load (MOL) approximation to explore the properties of the system, and develop a square-root-type rule to set the ceiling in the homogeneous case. We also use simulation to determine an optimal ceiling when we cannot route individual calls precisely. We propose approximations that may be computed for any call duration distribution, and compare their answers to exact differential-equation procedures for Exponential call durations.

Recent researches present that packet-mode scheduling may provide overall performance advantages over cell-mode scheduling in input queueing switches. But one significant drawback of existing packet-mode schedulers is that they tend to result in longer waiting time of short packets, whose delay is critical to the performance of IP networks. To mitigate the problem, this paper proposes a simple but effective packet-mode scheduling algorithm called Short Packets First (SPF), studies the corresponding buffering architecture, and evaluates the performance on throughput and packet delay. SPF achieves 100% throughput under an accurate Internet traffic model. Theoretical analysis proves that SPF can reduce the average packet waiting time of short packets as well as overall packets, under the condition of uniform Poisson arrival process and low to medium offered load. Furthermore, extended simulation studies show that SPF is superior to general packet-mode and cell-mode scheduling, especially under heavy offered load.

We propose a wireless access mechanism for web traffic packets in an integrated wireless code-division multiple-access system that has both voice and Web traffic. The proposed scheme is a medium-access control layer/link layer (MAC/LL) scheduling algorithm that consists of a two level control: admission control and packet scheduling. The admission control restricts the number of users in the system such that quality-of-service requirements [target signal-to-interference ratio (SIR) and delay] for both voice and Web traffic can be met. The packet scheduling balances the system interference on a slot-by-slot basis such that the target SIRs can be achieved for all users (voice and Web browsing sessions) with a higher scheduling priority for voice. Designing admission control for Web users based on the average offered rate per session is difficult due to the high variations in the offered load generated by heavy tailed distributions for Web traffic. To overcome this problem, we propose an admission control algorithm that adaptively estimates the aggregate average load based on load measurements using a sliding observation window.

An input and output buffering ATM switch conventionally operates in either Queueloss mode or Backpressure mode. Recently, a new mode, which is called Hybrid mode, was proposed to overcome the drawbacks of Queueloss mode and Backpressure mode. In Hybrid mode, when both the destined output buffer and the originfted input buffer are full, a cell is dropped. This thesis analyzes the cell loss rate and the cell delay of Queueloss, Backpressure and Hybrid modes in a switch adopting output-port expansion scheme under uniform traffic. Output-port expansion scheme allows only one cell from an input buffer to be switched during one time slot. If several cells switch to a same destined output port, the number of maximum transfer cells is restricted to K (Output-port expansion ratio). The simulation results show that if an offered load is less than 0.9, Hybrid mode has lower cell loss rate than the other modes; otherwise, Queueloss mode illustrates the lowest cell loss rate, which is a different result from previous researches. However, the difference between Hybrid and Queueloss modes is comparably small. As expected, the average cell delay in Backpressure mode is lower than those of Queueloss mode and Hybrid mode, since the cell delay due to the retransmission of higher number of dropped cells in Backpressure mode is not considered.

RT-MAC is a simple, elegant, and robust medium access control (MAC) protocol for use in transmitting real-time data in point-to-point ad hoc wireless local area networks (WLANs). Our enhancement of IEEE 802.11, real-time MAC (RT-MAC), dramatically reduces missed deadlines and packet collisions while increasing throughput by selectively discarding packets and sharing station state information. For example, RT-MAC is able to successfully transmit 40 2-way voice conversations in addition to a normalized offered load of 80%. IEEE 802.11 is only able to transmit 10 2-way conversations under the same conditions. In another instance, RT-MAC reduced packet collisions from 50% to less than 15% while increasing throughput by more than 50%. Herein we extend our previous performance analysis of RT-MAC to include integrated voice-data transmission. Stations using RT-MAC are interoperable with stations using IEEE 802.11 and show a significant performance improvement even when a minority of stations in the network employ RT-MAC.

This research is an analysis of the mixed system of data packets and CDMA voice signals transmitted by the channel load sensing protocol. The voice signal is a medium requiring instantaneous action to complete the transmission within a certain time whereas the data packet is a medium requiring high reliability but allows temporal delay. The blocking probability and Erlang capacity are derived for the voice signal while the throughput is derived for the data packet. The characteristics in the system using CDMA strongly depend on the number of simultaneously transmitting stations. In particular, in the present system, due to the mixture of different media, the effect of traffic on each medium needs to be considered. In regard to this point, this paper analyzes the effect of traffic on each medium from the relationship of the required quality of the media and the number of simultaneously transmitting stations. Then the optimization of the system with two kinds of media is considered. Further, the system capacity is derived as a guideline for the system optimization. The data offered load is derived to make maximally efficient use of the channel. © 2000 Scripta Technica, Electron Comm Jpn Pt 2, 83(4): 61–70, 2000

We provide an analytical proof that the departure rate of a CBR flow at an overloaded link with FIFO buffers is proportional to the flow's share of the total offered load at the link. This property of FIFO scheduling was recently validated in in a series of traffic measurement experiments. An extension of the analysis to a multi-node scenario shows that the output rate of a flow in a network with many overloaded FIFO switches approaches the pessimistic values given by blind multiplexing.

The Multimedia Cable Networks Systems (MCNS) Data Over Cable Service Interface Specifications (DOCSIS) is established as the primary cable network data communications standard. The head-end scheduling algorithm is not defined within the standard but it is the key function for providing the required performance capability. Computer simulation, using the Common Simulation Framework (CSF) 12 version of the CableLabs DOCSIS 1.0 computer model, has been used to predict the upstream system throughput and mean access delay. A prioritised first come first served scheduling algorithm has been considered to provide a baseline reference set of performance statistics against which other algorithms can be compared. The simulations, with theoretical confirmation, have shown that the maximum sustainable system throughput is 1965 kbps for a channel capacity of 2560 kbps and a packet size of 1500 octets, whereas for packet size of 100 octets the maximum sustainable throughput is only 1550 kbps. The mean access delay is found to vary between 10-900 ms depending on the offered load and assuming that the offered load does not exceed the capacity of the channel. Excess offered load causes service starvation according to the assigned priority.

We develop and analyze a simple, elegant medium access control (MAC) protocol for use in transmitting real-time data in point to point ad hoc wireless local area networks (WLANs). Our enhancement of IEEE 802.11, real-time MAC (RT-MAC), achieves dramatic reductions in mean delay, missed deadlines, and packet collisions by selectively discarding packets and sharing station state information. For example, in a 50 station network with a normalized offered load of 0.7, mean delay is reduced from more than 14 seconds to less than 45 ms, late packets are reduced from 76% to less than 1%, and packet collisions are reduced from 36% to less than 1%. Regression models are developed from simulation data to describe network behavior in terms of throughput, mean delay, ratio of late packets, and ratio of collisions. Stations using RT-MAC are interoperable with stations using IEEE 802.11.

This paper deals with planning system activities to support applications that have different contrasting requirements including timing constraints on tasks execution and correctness requirements. Our approach is based on a simple yet effective formulation of a value structure associated to the application tasks. Values are associated to each relevant outcome thus accounting for successful executions as well as for those which violate the application requirements. Moreover we assume degradable real time systems equipped with several execution techniques characterised by different execution costs and different levels of fulfilment of requirements (and associated reward). We propose an admission algorithm to select the part of the offered load to be executed, should overload occur. For all the admitted tasks the algorithm selects also the most suitable execution technique (among those available) to optimise the expected cumulated reward. We show that the algorithm provides the best solution to the optimisation problem resorting to the linear programming theory. Then we discuss the applicability of this result to systems operating in dynamic environments. A planner component is defined, responsible to collect information on the current status of the system and of its environment. The planner decides when a new ’plan‘ is required, and dynamically executes the admission algorithm to properly tune the usage of system resources.

In an Ethernet with high offered load, lengthy and highly variable packet delays can occur. These delays are due in part to the ''capture effect'' whereby a single station can dominate packet transmission on an Ethernet that uses the standard Binary Exponential Backoff (BEB) arbitration algorithm despite other stations also having packets queued for transmission. One proposed solution to this problem is the Binary Logarithmic Arbitration Method (BLAM) under review by the IEEE 802.3 standards committee. This paper presents a performance evaluation of BLAM and a new, simpler arbitration method called BEB2. It is shown that both BLAM and BEB2 reduce the mean and variance of packet delay and can thus improve the transport of real-time voice traffic on Ethernet.

We consider a loss system model of interest in telecommunications. There is a single service facility with N servers and no waiting room. There are K types of customers, with type i customers requiring A_i servers simultaneously. Arrival processes are Poisson and service times are exponential. An arriving type i customer is accepted only if there are R_i \,(\geq A_i ) idle servers. We examine the asymptotic behavior of the above system in the regime known as critical loading where both N and the offered load are large and almost equal. We also assume that R_1,\ldots, R_{K-1} remain bounded, while R^N_K\to\infty and R^N_K/\sqrt{N}\to0 as N \to\infty. Our main result is that the K dimensional “queue length” process converges, under the appropriate normalization, to a particular K dimensional diffusion. We show that a related system with preemption has the same limit process. For the associated optimization problem where accepted customers pay, we show that our trunk reservation policy is asymptotically optimal when the parameters satisfy a certain relation.

In this article, a generalized analytical model for the evaluation of different performance parameters of IEEE 802.15.6 standard is proposed. The standard is specifically designed for the purpose of supporting body-sensors, e-health monitoring and human body communication. The analytical framework of this standard is modelled using a three dimensional Markov chain with backoff stage, backoff counter and retransmission counter, as the stochastic parameters. To model the mechanism of backoff freezing, which occurs due to insufficient time period for data packet transmission in the allotted access period and the packet is dropped after certain recurrences of this incident, allocating the access period for the next data packet. Unlike previous analytical models, a dynamic or time-variant variable is incorporated in our proposed analytical model to support this mechanism. From the analytical model, we obtain the expressions for different performance metrics such as reliability or the probability of successful packet delivery, throughput and energy consumption. Using Monte-Carlo simulation we obtain the analytical results of reliability, throughput and energy consumption for varying offered load and payload, and the simulation results are obtained using Castalia simulator. The simulation results are compared with the analytical results of our proposed analytical model and an existing analytical model from present literature.

Power-load proportionality is a necessary feature in networks that aim at maximizing energy efficiency. Even in networks that handle near-capacity loads very efficiently, energy savings increase substantially if the power consumption closely follows the offered load. Rate adaptation is a common denomination for a set of technologies that operate at different timescales to establish power-load proportionality. To propel their deployment, it is useful to assess how compatible they are with existing network systems and identify the design upgrades that can maximize their energy savings in future networks. The formulation of accurate energy profiles for current equipment is a first step in this direction. We run extensive power measurement experiments to compile the energy profiles of five network systems from multiple vendors. Our results show only negligible signs of power-load proportionality in all five cases: to really make a dent in the carbon footprint and operational cost of packet networks, future system designs must pervasively deploy rate adaptation technologies, especially those that control power state transitions at the packet timescale.

Although the quasi-static scheduling based on Birkhoff-von-Neumann (BvN) decomposition can achieve high throughput with low operational complexity, its performance becomes less predictable when the input traffic is bursty. In this paper, we propose a deflection-compensated BvN (D-BvN) switch to enhance the performance. The D-BvN switch provides capacity guarantee for each virtual circuit (VC) by the BvN decomposition of average input traffic matrix, while copes with traffic burst by deflection. In particular, deflection scheme fully utilizes the spare capacity of starving VCs to deflect overflow traffic to other VCs and provide bandwidth for deflection traffic to re-access its desired VC. Analytical and simulation results show that it can achieve close to 100% throughput of offered load and the packet out-of-sequence probability is also negligible.

Issues of network design for the group virtual path (GVP) scheme, such as assignment of objective cell loss rates, algorithms for GVPs and congestion tolerance are discussed. The GVP is a previously proposed bandwidth sharing scheme. This scheme gives better statistical multiplexing efficiency since bandwidth is assigned to a group of virtual paths (VPs) rather than to an individual VP. Varying the cell loss rates does not have much effect on the acceptable offered load for low virtual channel (VC) peak speed. The cell loss rate for high VC peak speeds were evaluated by simulation and proved similar to virtual channel handlers (VCHs): switches and virtual path handlers (VPHs): cross connects. Therefore, objective cell loss rates should be assigned equally to VCHs and VPHs. Five algorithms are proposed for group VPs. The necessary transmission capacity given by each algorithm is slightly higher than optimum grouping but close to optimum groupings can be achieved by using these algorithms together. Congestion tolerance of the GVP scheme is better than that of the conventional VP scheme if only one VP is congested. If all VPs in a GVP are congested, congestion tolerance worsens. However, the transmission capacity necessary to compensate for congestion tolerance is smaller in the GVP scheme than in the conventional scheme.

For some distributed control applications, a highly reliable, high-speed time-critical communication network is required. Time-critical communication ensures completion of data transmission for an application, or that of transaction within a time specified by the application. A protocol has been presented for time critical communication that achieves highly effective traffic ratio of time critical data using two-level data queues. Estimates of effective traffic ratio are derived for the authors' protocol. Estimation parameters are input traffic load, ratio of time critical data in offered load, and time width for completion of data transmission.

We examine multiple independent sources providing status updates to a monitor through a first-come-first-served M/M/1 queue. We formulate a status-age timeliness metric and find the region of feasible average status ages for a pair of updating sources. In the presence of interfering traffic with a given offered load, we show the existence of an optimal rate at which a source should generate its updates.

The use of multimedia applications is increasing every year. These trends include the deployment using wireless local area network (WLAN) that allows users to access various services. The most significant thing in the use of wireless networks is the delay, collusion, attenuation and interference. In addition, it is difficult to provide quality of service (QoS) in WLAN as the network capacity also varies with the offered load [1]. There are several problems in QoS support for multimedia applications. For real-time multimedia applications, packet loss and packet drop due to excessive time delay is a major factor affecting the quality perceived by the user. For video streams, delay is a very important factor to consider when transmitting the entire video frame from the sender to the client. Level of losses incurred due to packet delay depends on the delay data transmitted in the form of a video stream. Video streaming applications usually impose an upper limit (threshold) packet loss. In particular, packet loss ratio required to be kept below the threshold for achieving an acceptable visual quality. Packet loss ratio is high due to network congestion or network traffic is very high (overload) that cause degradation of the quality of multimedia, especially in wireless networks with limited bandwidth. To reduce the rate of degradation is the necessary framework of adaptive mechanisms in the wireless network so that video streaming applications such as teleconferencing can work well.

In this paper, we propose a hybrid division cooperative transmission (HDCT) scheme for system throughput enhancement in enterprise femtocell systems. In HDCT scheme, a cooperative transmission (CT) scheme is applied to user equipments (UEs) located at the edge of a serving femtocell base station (fBS), whereas conventional scheme is applied to UEs located near the fBS. In the zone where CT is applied, a UE is capable of receiving desired signal from serving fBS as well as from an adjacent fBS. Thus, the UE achieves an improved signal to interference plus noise ratio (SINR) by using two time synchronized signals from two adjacent fBSs. Performance of the proposed scheme is evaluated in terms of call level quality of service (QoS) (i.e., call blocking probability) and packet level QoS (i.e., outage probability). We first measure the resource utilization and call blocking probability for the downlink channel for various offered load. Based on that, we obtain the outage probability and effective throughput of the system by computer simulations. Finally, the results of the simulation yield the optimum hybrid zone for achieving the maximum system throughput.

In order to verify the performance of aviation VHF Data Link Mode 2(VDL2) system that will be used in the next generation civil aviation in China, the physical layer and link layer models of VDL2 system based on OPNET are built first used for parameter simulation. Then, the effect of aircraft number on the average retransmission delay is simulated. The effects of offered load on the average subnet delay, the probability of successfully sending data packet, and the system throughput are further testified. The simulation results demonstrate that if the offered load is no more than 0.65, VDL2 system subnet delay can meet the requirement, throughput is raised significantly with the offered load increasing and the probability of successfully sending packet can reach to 80 percent. The results can be taken as an important reference to future practical application of VDL2 system.

The fiber distributed data interface (FDDI) standard supports synchronous and asynchronous data transmissions. It allows each station to have multiple classes of asynchronous data, and meets the requirements of different classes by means of a timer-based priority scheme. The performance of a FDDI network carrying multiple classes of traffic is considered. An analytical model is presented to evaluate the throughput of synchronous traffic and asynchronous traffic. The model can be used to evaluate the throughput of individual priority classes and the mean token-cycle time of the network, when the network offered load varies from very low values to high values. The governing equations for the throughput characteristics and mean token-cycle time are strictly functions of network parameters. Simulations are used to verify the analytical model. Simulation results are also used to examine the variation of the mean access delay characteristics of various classes and the token-cycle time distribution of the priority scheme. Limitations of the model are outlined. >