Energy efficiency is an important aspect in wireless networks and power management is a technique that can save energy. We investigate the power management in the IEEE 802.11 based infrastructure mode wireless LANs and seek the optimal power management parameters that can achieve good energy efficiency without degrading other performances. We propose two models for the power management scheme: the M/G/1 queue with bulk service model and the D/G/1 queue model. From the models, we obtain the analytical results for the energy efficiency and the response time performance metrics, which are controlled by a listen interval. Our simulation results are in good agreement with the analysis. Based on the analytical models and simulation, we propose to select the largest listen interval with the satisfaction of response time requirement. The analytical models can be used to predict system performance if system configurations are given. This can help system deployment and system optimization.

Location updates (LU) are emergency 911 procedures mandated by the FCC for cellular networks today; helping locate mobiles within 100 meters of their vicinity. This requires paging all mobiles within a vicinity regularly thereby leading to an increased use of bandwidth. This paper analyzes the existing schemes of hash based paging in LU procedures using Bloom filters (BF) and introduces two new schemes to improve performance: optimization of Bloom filters (OBF) and cumulative Bloom filters (CBF). An identifier field in the paging message is coded by applying hashing functions to create a BF and this is used to page a number of mobiles concurrently. False LU are the mobiles that may not belong to a particular paging area but still respond with LU updates. We observe that these false probabilities are very small and can be traded-off with the bandwidth gain. The results obtained compare the analytical and simulation results and their observation leads us to the goal of this research: to obtain a multi-fold increase in bandwidth gain at the cost of keeping the false positives to a realistic minimum

In this paper, we present a new energy-efficient bandwidth allocation scheme for wireless networks. First of all, we investigate the intrinsic relationship between the energy consumption and transmission rates of mobile terminals, in which transmission rate is determined through channel allocations. Then, we propose two schemes for connection admission control: victim selection algorithm (VSA) and beneficiary selection algorithm (BSA) with the intent to reduce energy consumption of each terminal. Moreover, we introduce an adjustment algorithm to statistically meet the demands for quality of service (QoS) during the resource allocation. The performance of the proposed schemes is evaluated with respect to energy consumption rate of each successfully transmitted bit, throughput and call blocking probabilities. An extensive analysis and simulation study is conducted for Poisson and self-similar, multi-class traffic.

UMTS (Universal Mobile Telecommunications System) uses WCDMA (wideband code division multiple access) as its radio interface. The main advantage of this is its flexibility in resource management. The paper investigates the performance of two main call admission control schemes that exploit this flexibility - the wideband power based (WPB) scheme and the throughput based (TB) scheme. It is observed that the WPB scheme works well when the UMTS network carries more voice traffic than data traffic and the TB scheme performs better when more data traffic is prevalent. Based on these results, the paper introduces a new scheme called the 'adaptive call admission control' (ACAC) based scheme. All three schemes are applied to a UMTS network and two quality of service (QoS) parameters are observed, the blocking and dropping probabilities. The system is modeled as a multi-rate, multi-class, priority based system. Confidence intervals are obtained that show that the ACAC scheme is the best in a heterogeneous UMTS environment.

Battery capacity of mobile terminals and radio bandwidth are both limited and precious resources in wireless networks. In this paper, we present a thorough performance study of energy-based admission control scheme to make the best use of these two resources for effective mobile communications. In order to reduce energy consumption of each terminal, we introduce a victim selection algorithm (VSA) and a beneficiary selection algorithm (BSA) for acquiring bandwidth and releasing bandwidth, respectively. To avoid potential compromise of quality of service (QoS) due to the concern of energy consumption in connection admission control, we further propose an adjustment algorithm to statistically meet the demands for QoS. The performance of the proposed schemes is evaluated with respect to energy consumption rate of each successfully transmitted bit, throughput and call blocking probabilities for a variety of traffic such as Poisson and self-similar, multi-class traffic.

Energy efficiency is an important aspect in wireless networks. We investigate the power management in the IEEE 802.11 and propose an enhanced power management scheme for the infrastructure mode network to improve energy efficiency. In our scheme, we set up multiple queues in an AP (Access Point) buffer, one queue for each station. The AP schedules the transmission sequence of buffered packets by piggybacking a bit map in TIM (Traffic Indication Map) message or data packets. Also, we propose to use the Queue-by-Queue algorithm to schedule the traffic in the AP. Our proposed power management scheme improves energy efficiency without degrading the response time of the system. Our simulation results are in good agreement with the analysis.

Given the wide and rapid deployment of "visitor networks" (Dory Leifer 2002), how to authenticate the user and account the usage on the per-packet basis securely and yet efficiently is still a challenging problem. In this paper, we explore the tradeoff between performance and security, and propose a per-data-packet authentication and access control protocol called RBWA (random-bit window-based authentication). Deployed in the IP layer, RBWA can work with various underlying linker layer specific mechanisms and network topologies. And compared with IPSec, it dramatically reduces the overhead and power consumption by adding only a few bits to each data packet. Furthermore, RBWA is strong against a suite of attacks such as replay attack, denial-of-service attack and spoofing etc. In particular, a robust antireplay window scheme is developed to counter the severe packet reordering. The performance of RBWA is evaluated via the simulation.

The set of TCP congestion control algorithms associated with TCP-Reno (e.g., slow-start and congestion avoidance) have been crucial to ensuring the stability of the Internet. Algorithms such as TCP-NewReno (which has been deployed) and TCP-Vegas (which has not been deployed) represent incrementally deployable enhancements to TCP as they have been shown to improve a TCP connection's throughput without degrading performance to competing flows. Our research focuses on delay-based congestion avoidance algorithms (DCA), like TCP-Vegas, which attempt to utilize the congestion information contained in packet round-trip time (RTT) samples. Through measurement and simulation, we show evidence suggesting that a single deployment of DCA (i.e., a TCP connection enhanced with a DCA algorithm) is not a viable enhancement to TCP over high-speed paths. We define several performance metrics that quantify the level of correlation between packet loss and RTT. Based on our measurement analysis, we find that, although there is useful congestion information contained within RTT samples, the level of correlation between an increase in RTT and packet loss is not strong enough to allow a TCP-sender to improve throughput reliably. While DCA is able to reduce the packet loss rate experienced by a connection, in its attempts to avoid packet loss, the algorithm reacts unnecessarily to RTT variation that is not associated with packet loss. The result is degraded throughput as compared to a similar flow that does not support DCA.

Given the wide deployment of IPSec/VPN (virtual private networks) technology, there might be a redundancy in security protection in some configurations. Various commercial companies have replaced 802.11 security with IPSec/VPN to protect the wireless LAN (local area network). How to do it in an efficient and lightweight way is a challenging research problem. This paper introduces a new lightweight identity authentication protocol, SOLA (Statistical One-bit Lightweight Authentication), for access control well suited for IEEE 802.11 networks with IP connections. This protocol prevents unauthorized access on a per packet basis. Since SOLA only adds one identity bit to each packet it will have a low impact on the network bandwidth and power consumption. The performance and efficiency of the SOLA protocol together with IEEE 802.11 is analyzed and evaluated via simulation.

Many corporate WAN architects are considering migrating from costly leased line circuits to 'private IP' services. In order to do so IP service providers must provide service level agreements (SLA) that offer robust assurances that cover service availability and performance. The industry direction is to model IP SLA after those offered by frame relay networks. The reality however is that the SLA surrounding today's private IP services only loosely mirror frame relay SLA. In this paper, we examine the intent and limitations of current IP SLA. We explore the feasibility of two significant enhancements. First we study the feasibility of reducing the time scales associated with the performance assessment to something that is meaningful to the corporate end user. Second we explore the potential benefits of extending current performance assessment methods with application level performance metrics. We present preliminary results suggesting that it is possible to offer performance assurances based on carefully chosen latency objectives as long as the average utilization of the access link is less than 50% over time scales identical to those used by the performance metrics. We introduce a Web-oriented performance metric (the Web Response Time metric) and show promising benefits over traditional ping based metrics. However, for either ping or Web-based performance metrics, due to the inherent nature of a best-effort IP service, IP performance assurances are only practical when applied over long time scales.

In the Internet era, there is a need to understand how traditional public switched telephone network (PSTN) services and the new Internet services can coexist. The best-effort based IP network will have to maintain the level of service that customers from the PSTN world expect. This study contributes to the traffic engineering of such a system by presenting an analytical model for integrating voice over IP (VoIP) traffic with multi-fractal data traffic. This model is based on the stochastic fluid flow model. Effects on delay and loss performance of adding VoIP traffic to a data link as well as capacity requirements to maintain a certain quality of service are discussed.

We investigate the performance modelling of a packet routing server (PRS) architecture using the dynamic synchronous transfer mode (DTM) technology via simulation and the derivation of analytical models for the system's throughput, average queuing delay, and average queue length. Our results demonstrate that our analytical models achieve a good approximation in comparison with the simulation results.

Wireless systems have been designed as microcellular architectures in order to provide higher capacity. This reduced coverage area of a cell has caused a higher rate of hand-off events, and providing continuous service to a wireless call after hand-off has been an important issue. A simple and efficient bandwidth reservation scheme is proposed, which dynamically adjusts the amount of bandwidth reserved for hand-off calls in a cell, based on the amount of cellular traffic in its neighboring cells. The performance of the proposed scheme is evaluated and compared with the guard channel scheme which is widely used in current cellular networks.

Our research focuses on end-to-end congestion avoidance algorithms that use round trip time (RTT) fluctuations as an indicator of the level of network congestion. The algorithms are referred to as delay-based congestion avoidance or DCA. Due to the economics associated with deploying change within an existing network, we are interested in an incrementally deployable enhancement to the TCP/Reno protocol. For instance, TCP/Vegas, a DCA algorithm, has been proposed as an incremental enhancement. Requiring relatively minor modifications to a TCP sender, TCP/Vegas has been shown to increase end-to-end TCP throughput primarily by avoiding packet loss. We study DCA in today's best effort Internet where IP switches are subject to thousands of TCP flows resulting in congestion with time scales that span orders of magnitude. Our results suggest that RTT-based congestion avoidance may not be reliably incrementally deployed in this environment. Through extensive measurement and simulation, we find that when TCP/DCA (i.e., a TCP/Reno sender that is extended with DCA) is deployed over a high speed Internet path, the flow generally experiences degraded throughput compared to an unmodified TCP/Reno flow. We show (1) that the congestion information contained in RTT samples is not sufficient to predict packet loss reliably and (2) that the congestion avoidance in response to delay increase has minimal impact on the congestion level over the path when the total DCA traffic at the bottleneck consumes less than 10% of the bottleneck bandwidth.

In order to provide a ubiquitous telecommunications network which merges the concepts of wireless mobile communications with the use of ATM as a transport mechanism, this paper focuses on the problem of reducing the effects of the high bit error rate on the data transmission. We present a simulator which can be used to study the effects of various coding techniques on the overall system bit error rate. Among these coding techniques are convolutional coding, interleaving, fragmentation, and puncturing. We show that convolutional coding, along with interleaving or fragmentation, improves transmission reliability over the bursty error channel.

To understand the performance of MPEG-1 videos when transmitted through B-ISDN ATM networks, this paper analyzes an equivalent simulated network. Allocation of network bandwidth is based on equivalent capacity with MMBP inputs providing background traffic. Cell losses are examined and found to be near target. Also, an analysis of cell delay indicates that delay distributions can be approximated by gamma distributions and that frame-to-frame cell delay is uncorrelated for the test network. Finally, a method to predict MPEG-1 decode buffer underruns is developed and validated.

The aauthors present a performance study of a digital mobile communication system (DMCS) which is a two-way radio frequency data communication network. The purpose of this modeling (multi-server queueing) effort is to provide the mean inbound and outbound message delays under different traffic intensities. Due to the acknowledgement and retransmission mechanism, the inbound and outbound traffic interact closely with each other and the system behavior becomes complicated. Through this study, they are able to identify the operational bottleneck of the DMCS and obtain the system capacity under which the network provides the users with a satisfactory communication service.>

A queueing model is analyzed for systems with a hierarchy of storage where the access time to all the stored images is not the same. A typical PACS consists of a database of medical images, display stations, acquisition systems, and a communication medium. The proposed model is for the delay analysis of the image database, the communication channels, and the picture viewing stations (PVSs). Performance curves are presented for the delay in retrieving images from the database, in the transmission of images, and in processing them in the image processor. The mean delay (that is, the time it takes for a radiologist to view a medical exam from the moment that a request packet is sent to the image database till the time the images are received) is the sum of all the delays mentioned above. The distributions of these delays can be easily evaluated and plotted by using the inverse of the Laplace transforms. From the curves provided, the end-to-end delay for various amounts of load in the system can be accurately estimated.>

Interconnected heterogeneous networks are presented and their performance analyzed. The interconnected networks consist of token ring, fiber distributed data interface (FDDI), and carrier sense multiple access with collision detection (CSMA/CD) networks, each of which connects a large population of users. In the interconnected networks, the local users on the token ring send intra- or internetwork packets. The internetwork packets are transmitted to the CSMA/CD network via gateways and the FDDI network. In the meantime, the local users on the FDDI and CSMA/CD networks generate intranetwork packets. Since the maximum packet sizes of these networks are different, packet fragmentation and reassembly are sometimes necessary. These functions are performed by gateways. The model focuses on the end-to-end delay between two end users on the token ring and CSMA/CD network. Analytic models for calculating the delays for the FDDI network and the interconnected networks are presented, and the measurements from the models are compared with that from the simulation models, validating the analytic models.>

The topology described applies to any general-purpose computer networking environment. The scheme is named the spiral topology because the architecture is built around modules of four computer nodes each, connected by top and bottom spirals. The spiral technology features a simple internal self-routing algorithm that adapts quickly and automatically to failed nodes or links. The six most important direct consequences of the network architecture are the topology's: (1) ease of expansion; (2) fast, on-the-fly self-routing; (3) extremely high tolerance to network faults; (4) increased network security; (5) potential for the total elimination of store-and-forward transmissions due to routing decision delays; and (6) rendering the maximum path length issue moot.>

This paper presents an analysis of the performance of a digital speech interpolation (DSI) system as a function of the traffic offered to the system. We assume that the DSI overload degradation occurs as clipping or freezeout. This is identical to TASI freezeout. Unlike conventional analysis, which assumes all of the trunks entering the DSI system are busy all the time, we calculate the freezeout fraction as a function of the traffic offered. In almost all practical situations the freezeout fraction, computed based on traffic assumptions, is considerably less than that computed assuming all trunks are busy. When a DSI system is used to compressNtrunks down intocchannels, there is always a tradeoff between the blocking probability in the trunk group and the degradation introduced by the DSI system. This tradeoff depends on the traffic offered and is explored in this paper.

A theoretical method of evaluating degradations of variable rate coders in a multichannel digital speech interpolation (DSI) system is developed. Each of the coder outputs has a variable rate based on the algorithm. The DSI system multiplexes the outputs of these variable rate coders into a fixed bit rate channel. During periods of high activity all active users are served, but at a reduced rate depending on the demand. The degradation due to high activity is shared by all active users. This system avoids speech clipping and "freeze-out" distortion. Theoretical expressions of the system overload probability and the probability of degradation to a particular user in the DSI system are derived. Two types of variable rate coders, namely, a constant quality subband coder and a constant noise subband coder, are chosen and used as examples. Comparisons of the degradations are made between the theoretical results and computer simulated results for the two types of variable rate coders, and close agreement is observed. The theory is applicable to other variable rate coding algorithms as well. In this study, all of the simulations are made at 40 percent speech activity and the average rate of the variable rate coders is close to 16 kbits/s. Objective quality measures indicate that in a system with a trunk size larger than 40, the variable rate coder DSI system can achieve a 2:1 compression with a degradation of less than 1 dB compared to non-DSI variable rate coders. This corresponds to a total gain of 8:1 when compared to 64 kbit/s PCM.

In the correspondence1the author presented an approximate formula for the average failure rate in a periodically repaired n - 1/n parallel redundant system. It was assumed that the system consists of n parallel identical elements, each of which has a constant failure rate λ. The system is also repaired every T hours. It was shown that an approximate expression for the average failure rate of such a system is n(n - 1) λ2T/2. Upper and lower bounds were also given. In this comment a closed form exact expression for the average failure rate is presented.

A tandem queue model is developed that models the end-to-end delay behavior in networks that employ sliding window flow control. Messages arriving to find the window filled are assumed to be queued outside the network to await their turn to enter. The model is analyzed using a hierarchical decomposition method; the key step entails incorporating the fact that the interdeparture process is not exponentially distributed. End-to-end delay in virtual circuits can then be obtained by modelling a virtual route as a tandem queue [1] using the method of “adjusted rates.”

The problem of fmdlng those optimum scheduling algorithms for time-shared systems that mlmmize a cost function that depends on waiting time and required service time IS considered An optimality condmon which sometimes leads to infeasible algorithms is established The procedure is unproved upon by use of a mathematical programming technique but still does not always generate feasible algorithms.These results are used as upper bounds on the performance of known feasible algorithms so that it is possible to evaluate how close to optimal the present algorithms come.

By classifying traffic conditions into four categories, we investigate the preference for a deterministic or an adaptive routing strategy with respect to a traffic category. The four categories are: 1) balanced, emulating known and stationary traffic conditions; 2) balanced with surge, emulating a balanced traffic condition with possible unexpected sudden increases in traffic demands between some source-destination pairs; 3) unbalanced, emulating unknown or nonstationary traffic conditions with low to moderate traffic loading; and 4) chaotic, emulating unknown or nonstationary traffic conditions with heavy traffic loading. Our study concludes that a well-chosen deterministic strategy is better for balanced traffic; both deterministic and adaptive procedures are appropriate for balanced traffic with surge; and a well-chosen adaptive strategy is preferred for the unbalanced and chaotic conditions. We also attempt to define a routing strategy that behaves almost deterministically when appropriate and adaptively otherwise. We contend that an adaptive routing strategy with a second-order metric1 + a_{1}Q + a_{2}Q^{2}, where1 \gg a_{1} \gg a_{2}are constants andQis the queue size at a link, is a good compromise. (Values of a1and a2depend on network topology and congestion control strategies.) Our reasoning and results are supported and verified by experimenting with a detailed simulation program.