Search

PDF file - 128K

Entry into postgraduate medical training in Canada is facilitated through a national application and matching system which establishes matches between applicants and training programs based on each party's stated preferences. Health human resource planning in Canada involves many factors, influences, and decisions. The complexity of the system is due, in part, to the fact that much of the decision making is dispersed among provincial, territorial, regional, and federal jurisdictions, making a collaborative national approach a challenge. The national postgraduate application and matching system is one of the few aspects of the health human resources continuum that is truly pan-Canadian. This article examines the evolution of the application and matching system over the past half century, the values that underpin it, and CaRMS' role in the process.L’entrée dans la formation médicale postdoctorale au Canada est facilitée par un système national de demande et de jumelage qui établit des correspondances entre les candidats et les programmes de formation en fonction des préférences déclarées par chaque partie.La planification des ressources humaines en santé au Canada implique de nombreux facteurs, influences et décisions. La complexité du système est due, en partie, au fait qu'une grande partie du processus décisionnel est réparti entre les compétences provinciales, territoriales, régionales et fédérales, ce qui rend difficile une approche nationale collaborative. Le système national de demande postdoctorale et de jumelage est l'un des rares aspects du continuum des ressources humaines en santé qui est véritablement pancanadien.Cet article examine l'évolution du système d'application et de jumelage au cours du dernier demi-siècle, les valeurs qui le sous-tendent et le rôle du CaRMS dans le processus.

To develop a French version of the Overt Behaviour Scale (OBS) and examine some of its psychometric properties.The scale was adapted and validated according to standard guidelines for cross-cultural adaptation of questionnaires (Échelle des comportements observables; ÉCO). The reliability and construct validity of the ÉCO were studied among 29 inpatients and outpatients who sustained an acquired brain injury. The instruments were administered by 12 clinicians located at eight rehabilitation centres and the local brain injury association.The ÉCO provided behaviour profile descriptives much like the original scale. It showed excellent reliability and good convergent and divergent validity, as reflected by significant associations with other measures that contained similar behavioural items and by the absence of signification correlations with broader constructs such as physical and cognitive abilities.This study provides evidence that the ÉCO behaves much like the original OBS, has promising initial findings with respect to reliability and validity and is a valuable research and clinical instrument to assess the severity and typology of challenging behaviour after an acquired brain injury and to monitor the evolution of behaviours after intervention in French and bilingual communities.

This article describes the work performed by ITU-T SG15Q13 for defining the first telecom profile based on the use of IEEE Std 1588-2008. The first profile is specifically developed for the distribution of frequency using unicast IPv4 transmission, and required adaptation of the IEEE1588 protocol to make it suitable for the telecom environment. The objectives, reasons, and results of this adaptation are explained in this article. Since the distribution of phase/time is also gaining importance in telecom, the article briefly discusses the objectives, reasons, and upcoming work for the definition of another profile that will leverage other functions and clocks defined in IEEE1588.

During our daily displacements, we should consider the individuals advancing toward us in order to avoid a possible collision with our congeneric. We developed an experimental design in a virtual immersion room, which allows us to evaluate human capacities for avoiding collisions with other people. In addition, the design allows participants to interact naturally inside this immersive virtual reality setup when a pedestrian is moving toward them, creating a possible risk of collision. Results suggest that the performance is associated with visual and motor capacities and could be adjusted by cognitive social perception.

One of the new challenges associated with full-duplex Ethernet is that of providing for network congestion control. The IEEE 802.3x Standard does not specify the switch architecture for implementing PAUSE flow control or at what point a MAC Control entity actually generates a PAUSE frame. This is an implementation issue that is product specific and as a result many different switch architectures are possible. There are also a number of limitations of the PAUSE flow control mechanism when implemented in Ethernet switches. These issues have not been adequately addressed in the literature. In addition, multimedia traffic such as real-time voice and streaming video are now being deployed over switched Ethernet networks, thus calling for congestion control with service differentiation for the various classes of traffic. Here we examine current Ethernet switch architectures and show that the PAUSE flow control when implemented in these architectures does not provide service selectivity and differentiation, making it unsuitable for real-time traffic.

In our previous work [J. Aweya, M. Ouellette, D. Montuno, K. Felske, Rate-based proportional-integral control scheme for active queue management, International Journal of Network Management, John Wiley & Sons, Ltd., vol. 16, issue 3, May-June 2006, pp. 203-231] we argued that rate-based active queue management (AQM) schemes are most appropriate for high speed links which, typically, have small buffers in relation to the bandwidth-delay product of the link. We continue our discussion in this paper to study the performance of other rate-based AQM controllers. We consider integral (I) and proportional (P) rate-based controllers for the AQM problem, we then characterize all stabilizing feedback gains for the closed-loop TCP/AQM system. Using a network topology with short- and long-lived TCP flows, we present simulation results for the rate-based AQM controllers. We observe that the I-controller is able to control properly the system compared to the P-controller. The P-controller in most cases produces offsets from the target values. To further support our observations, we use a closed-loop control model and control theory to explain why the P-controller has major limitations in the control of a first-order plant like our TCP/AQM first-order plant.

This article discusses the evolving transport architecture, covering some of the synchronization distribution problems to many endpoints where mobile backhaul and TDM emulation occur. It shows how synchronous Ethernet fits into both the Ethernet and synchronization architectures, and discusses how this helped development in standardization bodies. Standardization allows the key building blocks of Ethernet silicon and specific timing devices to be developed, which allows a robust system implementation to be constructed while allowing interworking with and migration from existing SONET/SDH-based transport infrastructure. Finally, results are shown that indicate a very high level of performance is achievable with Synchronous Ethernet not subject to the normal packet delay variation and traffic load conditions that can occur in packet based networks.

In this paper we determine the stability bounds for the DRED active queue management (AQM) algorithm using a previously developed nonlinear dynamic model of TCP. We develop a second-order linear model with time delay by linearizing the nonlinear model. Using the Pade approximation of time-delayed system e-R0s, where R0 is the delay in the system, we then determine the range of stabilizing gains of DRED when controlling the second-order system with time delay R0. We also present examples showing the stability bounds of the DRED controller gain for networks with different parameters such as link capacity, load level, and round-trip time. In addition, we describe an efficient implementation of the DRED AQM algorithm.

Àdeux reprises dans les dernières années, j’ai eu l’occasion d’entendre des communications surmon oeuvre dans le cadre de colloques universitaires. Chaque fois, la sensation était étrange de se trouver dans une salle pour entendre un chercheur présenter le fruit de ses recherches surmon théâtre. C’était commeme regarder dans un miroir et découvrir de nouveaux aspects de mon visage.

Circuit emulation service (CES) allows time-division multiplexing (TDM) services (T1/E1 and T3/E3 circuits) to be transparently extended across a packet network. With circuit emulation over IP, for instance, TDM data received from an external device at the edge of an IP network is converted to IP packets, sent through the IP network, passed out of the IP network to its destination, and reassembled into TDM bit stream. Clock synchronization is very important for CES. This paper presents a clock synchronization scheme based on a double exponential filtering technique and a linear process model. The linear process model is used to describe the behaviour of clock synchronization errors between a transmitter and a receiver. In the clock synchronization scheme, the transmitter periodically sends explicit time indications or timestamps to a receiver to enable the receiver to synchronize its local clock to the transmitter's clock. A phase-locked loop (PLL) at the receiver processes the transmitted timestamps to generate timing signal for the receiver. The PLL has a simple implementation and provides both fast responsiveness (i.e. fast acquisition of transmitter frequency at a receiver) and significant jitter reduction in the locked state. Copyright © 2006 John Wiley & Sons, Ltd.

Accurate source clock recovery is an important element of circuit emulation services (CES) over packet networks. A well-known and widely implemented technique for clock recovery in CES is the one that is based on packet inter-arrival time (sometimes called time difference of arrival (TDOA)) averaging. The technique is very simple to implement but provides good performance only when packet losses and packet delay variation (PDV) are very low and well controlled. This technique has not been fully characterized analytically in the literature. In this paper, we provide a full analytical examination of this well-known clock recovery technique. We analyze the effects of correlation of the PDV experienced by the constant bit rate (CBR) traffic stream on the quality of the clock recovered by a receiver. We prove analytically that, for a general input process, high correlation of the PDV produces a large variance of the recovered clock. The paper also describes simple all-digital implementations of the clock recovery scheme using standard digitally controlled oscillators (DCOs).

Most high-speed links do not have adequate buffering and as a result Active Queue Management (AQM) schemes that utilize queue size information for congestion control cannot be effectively applied on these links. A high-speed link will, typically, have small buffers in relation to the bandwidth-delay product of the link. In this paper we argue that rate-based AQM schemes be used for such links. The goal here is to match the aggregate rate of the active TCP connections to the available capacity while maintaining minimal queue size and high link utilization. The AQM scheme described here employs a Proportional-Integral (PI) control strategy and explicitly takes into account the time delay in the control process.

In this paper, we present a clock synchronization scheme based on a simple linear process model which describes the behaviors of clocks at a transmitter and a receiver. In the clock synchronization scheme, a transmitter sends explicit time indications or timestamps to a receiver, which uses them to synchronize its local clock to that of the transmitter. Here, it is assumed that there is no common network clock available to the transmitter and the receiver and, instead, the receiver relies on locking its clock to the arrival of the timestamps sent by the transmitter. The clock synchronization algorithm used by the receiver is based on a weighted least-squares criterion. Using this algorithm, the receiver observes and processes several consecutive clock samples (timestamps) to generate accurate timing signals. This algorithm is very efficient computationally, and requires the storage of only a small number of clock samples in order to generate accurate timing signals.

The relative differentiated service model provides assurances for the relative quality ordering between service classes, rather than for the actual service level in each class. In this paper, we describe a relative loss rate differentiation scheme where packet drop probabilities are determined according to an active queue management (AQM) mechanism based on random early detection (RED) in a first-in first-out (FIFO) queue, are weighted in inverse proportion to the price that the network operator assigns to each service class. Basically, we describe a scheme where relative loss rate differentiation is incorporated directly into AQM. Most TCP flows today, particularly Web flows, can be characterized as short-lived flows. Using simulations with short-lived TCP flows, we show that the scheme is very effective in ensuring relative loss rate differentiation between service classes during times of network congestion. Copyright © 2004 John Wiley & Sons, Ltd.

The relative differentiated service model is one of several models proposed for service differentiation in networks [IEEE Network Sept/Oct (1999) 26]. In this model, an assurance is given that 'higher classes will be better, or at least no worse than lower classes.' This paper describes a relative loss rate differentiation scheme based on RED. The scheme is used for differentially dropping packets in a FIFO queue during times of congestion. The main idea is, if packet losses are unavoidable in the FIFO queuing system, then they should be distributed among the different service classes in the queue in inverse proportion to the service price or weight assigned to each class. The simulation studies using TCP traffic show that the scheme is very effective in ensuring relative loss rate differentiation between service classes.

Feedback delays arise in the control of a computer network--from the information transfer process itself and from the processing of control signals at the network nodes. Flow control of data sources in a computer network often results in a time-delayed control problem. Feedback delay reduces the stability of a system. In this paper we discuss how to use the Routh-Hurwitz stability criterion to design and analyze the stability of a flow control algorithm with feedback delay.

This paper presents exact stability analysis of a rate control algorithm described in Perform. Eval. 2001; 43(2–3):63–94; Int. J. Commun. Systems 2001; 14(6):593–618. The stability regions of the rate control process in the presence of control loop delay are analysed. The rate control process is represented by delay-difference equation and the criteria for asymptotic stability are derived in terms of the control parameters and control loop delay. The analysis shows that the approximate upper bound of the control gain derived in Aweya et al. is very close to the exact bound developed here. Using theoretical calculations performed in the discrete-time domain, we show that as the feedback time delay d increases, the intensity of control (i.e. the control gain α) must decrease in order for the system to remain stable. Copyright © 2004 John Wiley & Sons, Ltd.

Due to Ethernet's ubiquity, simplicity, scalability and cost effectiveness there is significant customer demand for Ethernet-based access and transport in the metropolitan network. Many service providers have recognized this need and are currently establishing Ethernet-based services to meet this demand. The migration to all-Ethernet access will not be instantaneous since many customers currently have legacy TDM access interfaces on their routers and PBX equipment. Circuit Emulation Services (CES) over Ethernet provides TDM circuit emulation to support TDM traffic such as T1/E1, T3/E3, OC3/12, etc. This two-part paper presents the application of CES over Ethernet as well as a new technology that addresses the issues associated with clock recovery and synchronization in an Ethernet network with its inherent network jitter. Part 1 describes a clock synchronization technique where a transmitter periodically sends explicit time indications or timestamps to a receiver to enable the receiver to synchronize its local clock to the transmitter's clock.

There is current interest in differentiated service architectures where packets with different priorities can share the same queue. In the case of congestion, packets marked with higher drop probability are preferentially dropped in order to make buffer room for packets marked with lower drop probability. Active queue management (AQM) based on randomized packet dropping has become a key component of this packet forwarding model. This paper extends our previously developed AQM algorithm called DRED with multiple packet drop precedence to allow for priority treatment of traffic in a network. The main advantage of using the DRED algorithm is the lower parameter configuration complexity it offers and the ease of configuration for a wide range of network conditions.

This paper describes a prototype implementation and experimental results for unstructured circuit emulation service (UCES) of T3 data stream over Ethernet. As explained in Part 1 of this paper, packet-switched networks such as Ethernet are not designed to transport TDM data and so have no inherent clock distribution and synchronization mechanisms. Thus, to allow the frequency of the source TDM stream to be regenerated at the receiver, the prototype employed the clock synchronization scheme described in Part 1 of this paper. Our experiments showed that the recovered clock conforms to ITU-T G.824 requirements even for networks that introduce high jitter and packet loss.

Explicit transmission control protocol (TCP) window control through the modification of the receiver’s advertised window in ACK packets is one of the ways intermediate network elements can contribute to the end-to-end TCP control. The TCP receiver’s advertised window, which indicates the level of the receive buffer of a TCP connection, limits the maximum window and consequently the throughput that can be achieved by the TCP sender. Thus, appropriate reduction of the advertised window by intermediate network elements can control the number of packets sent from a TCP sender. This paper describes a TCP window control scheme for a shared memory system with multiple queues. A dynamic buffer threshold, computed using a simple recursive algorithm, is used to dynamically allocate buffer space to the queues.

Most active queue management schemes maintain an average of the queue length which they use together with a number of queue thresholds to detect congestion. However, the setting of the queue thresholds is problematic because the required buffer size for good sharing among TCP connections is dependent on the number of TCP connections using the buffer. This paper describes an improved active queue management scheme which dynamically changes its threshold settings as the number of connections and system load changes. This technique allows network devices to effectively control packet losses and TCP timeouts while maintaining high link utilization.

A multiple (priority) queueing system allows a network node to manage the queueing of packets in such a way that higher priority packets will always be served first, low priority packets will be discarded when the queue is full, and for same-priority packets any interference between them will be prevented. This paper describes a TCP window control scheme for a shared memory device that has buffer memory logically organized into multiple queues. To handle changing queue traffic loads, the shared memory device uses a dynamic buffer threshold mechanism to allocate buffer space to the queues. The TCP window control scheme allows the receiver's advertised window size in ACK packets to be modified at the network queue in order to maintain the queue size at a computed dynamic threshold. Copyright © 2003 John Wiley & Sons, Ltd.

In explicit TCP rate control, the receiver's advertised window size in acknowledgment (ACK) packets can be modified by intermediate network elements to reflect network congestion conditions. The TCP receiver's advertised window (i.e. the receive buffer of a TCP connection) limits the maximum window and consequently the throughput that can be achieved by the sender. Appropriate reduction of the advertised window can control the number of packets allowed to be sent from a TCP source. This paper evaluates the performance of a TCP rate control scheme in which the receiver's advertised window size in ACK packets are modified in a network node in order to match the generated load to the assigned bandwidth in the node. Using simulation and performance metrics such as the packet loss rates and the cumulative number of TCP timeouts, we examine the service improvement provided by the TCP rate control scheme to the users. The modified advertised windows computed in the network elements and the link utilization are also examined. Copyright © 2002 John Wiley & Sons, Ltd.

The transmission control protocol (TCP) receiver's advertised window (i.e. the receive buffer of a TCP connection) limits the maximum window and consequently the throughput that can be achieved by the sender. Thus, the idea behind the technique generally known as 'TCP rate control' is to match the offered network load to the available resources by modifying at an intermediate network device, the receiver's advertised window in TCP acknowledgments returning to the sources. In this paper, we propose a new TCP rate control scheme for a shared buffer where the buffer is logically organized into multiple queues. In the scheme, a dynamic buffer threshold is used to ensure efficient and fair usage of buffer memory among the queues. Conventional schemes allocate buffer space to each queue through the use of static buffer thresholds. This can result in unnecessary packet drops which lead to poor network performance since congested or heavily loaded queues cannot gain access to buffers not utilized by lightly loaded queues.

A number of active queue management algorithms for TCP/IP networks such as random early detection (RED), stabilized RED (SRED), BLUE, and dynamic RED (DRED) have been proposed in the past few years. This article presents a comparative study of these algorithms using simulations. The evaluation is done using the OPNET Modeler, which provides a convenient and easy-to-use platform for simulating large-scale networks. The performance metrics used in the study are queue size, packet drop probability, and packet loss rate. The study shows that, among the four algorithms, SIZED and DRED are more effective at stabilizing the queue size and controlling the packet loss rate while maintaining high link utilization. The benefits of stabilized queues in a network are high resource utilization, bounded delays, more certain buffer provisioning, and,traffic-load-independent network performance in terms of traffic intensity and number of TCP connections.

In this paper, we propose a new active queue management scheme for a network device (e.g. router, switch, etc.) with a shared buffer where the buffer is logically organized into multiple queues. Conventional schemes allocate buffer space to each queue through the use of static buffer thresholds. Although very simple, these static thresholds are often determined using simple heuristics and cannot adapt to changing traffic conditions. These schemes could also result in unnecessary packet drops which lead to poor network performance since congested or heavily loaded queues cannot gain access to buffers not utilized by the lightly loaded queues. The proposed active queue management scheme, which uses dynamic buffer thresholds, adapts to changing load conditions and dynamically allocates buffer space to all queues using a simple recursive algorithm. The implementation of the scheme is also simple and requires low processing overhead to compute the dynamic buffer thresholds and packet drop probabilities.

This paper describes an overload control scheme for web servers which integrates admission control and load balancing. The admission control mechanism adaptively determines the client request acceptance rate to meet the web servers' performance requirements while the load balancing or client request distribution mechanism determines the fraction of requests to be assigned to each web server. The scheme requires no prior knowledge of the relative speeds of the web servers, nor the work required to process each incoming request.

In this paper, we evaluate the performance of TCP in a network where the IEEE 802.3x flow control and the ATM Available Bit Rate flow control mechanisms are interworked. The objective is to examine whether an interworking of the two flow control mechanisms can improve the performance of upper-layer protocols like TCP which is an end-to-end connection-based transport protocol. The simulation studies show that the two interworked flow control mechanisms result in high link utilization while providing very low loss rates when used in conjunction with TCP. The flow control mechanisms complement each other.

This paper describes a multi-level active queue management scheme that combines packet classification and random early detection (RED) capabilities to provide differentiated performance characteristics for different classes of service. The proposed RED scheme uses a simple control-theoretic algorithm to randomly discard packets with a load-dependent probability when a buffer in a router gets congested.

We describe in this paper a new TCP ACK pacing scheme that dynamically tunes its behavior to account for variations or changes in the network load. The scheme does not require the knowledge of when TCP is in the slow-start or congestion avoidance phase to determine the proper ACK pacing rate.

In this paper, we analyze quantitatively the stability and equilibrium performance of a rate allocation scheme. This kind of analysis is important because in computer networks where multiple applications compete and share limited amount of system resources, some applications tend to suffer from variations in resource availability, and are desired to adapt their behavior to the resource variations of the network. Also, we analyze the fairness properties of the rate allocation scheme. Fairness guarantees are provided without collecting or analyzing state information on individual flows. The rate allocation scheme is able to provide efficiently and effectively rate guarantees, fairness, responsiveness and robustness without having to incorporate numerous tuning parameters as is normally done in most schemes.

It is now widely accepted that a RED [2] controlled queue certainly performs better than a drop-tail queue. But an inherent weakness of RED is that its equilibrium queue length cannot be maintained at a preset value independent of the number of TCP active connections. In addition, RED's optimal parameter setting is largely correlated with the number of connections, the round-trip time, the buffer space, etc. In light of these observations, we propose DRED, a novel algorithm which uses the basic ideas of feedback control to randomly discard packets with a load-dependent probability when a buffer in a router gets congested. Over a wide range of load levels, DRED is able to stabilize a router queue occupancy at a level independent of the number of active TCP connections. The benefits of stabilized queues in a network are high resources utilization, predictable maximum delays, more certain buffer provisioning, and traffic-load-independent network performance in terms of traffic intensity and number of connections. Copyright © 2002 John Wiley & Sons, Ltd.

In TCP, congestion control as well as error recovery are implemented by a sliding window. The dynamics of TCP specifically, a mismatch between the TCP window and the bandwidth-delay product of the network can sometimes cause the network switches or routers to accumulate large queues, resulting in buffer overflows, reduced throughput, unfairness and underutilization. It is generally accepted that there is a limit to how much control can be accomplished from the congestion control mechanisms in the end systems. Some mechanisms are needed in the intermediate network elements to complement the endpoint congestion avoidance mechanisms. We describe in this paper, a new TCP rate control scheme based on a simple recursive algorithm. The idea behind the algorithm is to match the offered network load to the available resources by modifying at an intermediate network element, the receiver’s advertised window in TCP acknowledgments returning to the source. We show through simulations that the scheme can efficiently control TCP traffic to limit queue buildups and buffer requirements at the network nodes, resulting in significant improvements in delay, packet loss rates, fairness in the distribution of the maximum achievable window size, and link utilization.

The basic idea behind active queue management schemes such as random early detection (RED) is to detect incipient congestion early and to convey congestion notification to the end-systems, allowing them to reduce their transmission rates before queues in the network overflow and packets are dropped. The basic RED scheme (and its newer variants) maintains an average of the queue length which it uses together with a number of queue thresholds to detect congestion. RED schemes drop incoming packets in a random probabilistic manner where the probability is a function of recent buffer fill history. The objective is to provide a more equitable distribution of packet loss, avoid the synchronization of flows, and at the same time improve the utilization of the network. The setting of the queue thresholds in RED schemes is problematic because the required buffer size for good sharing among TCP connections is dependent on the number of TCP connections using the buffer. This paper describes a technique for enhancing the effectiveness of RED schemes by dynamically changing the threshold settings as the number of connections (and system load) changes. Using this technique, routers and switches can effectively control packet losses and TCP timeouts while maintaining high link utilization.

This paper describes the Dynamic-RED (DRED) algorithm, an active queue management algorithm for TCP/IP networks. In random early detection (RED), one of the goals is to stabilize the queue lengths in routers. However, the current version of RED does not succeed in this goal because the equilibrium queue length strongly depends on the number of active TCP connections. Using a simple control-theoretic approach, DRED randomly discards packets with a load-dependent probability when a buffer in a router gets congested. Over a wide range of load levels, DRED is able to stabilize a router queue occupancy at a level independent of the number of active TCP connections. The algorithm achieves this without estimating the number of active TCP connections or flows and without collecting or analyzing state information on individual flows. The benefits of stabilized queues in a network are high resources utilization, bounded delays, more certain buffer provisioning, and traffic-load-independent network performance in terms of traffic intensity and number of connections.

To help improve the performance of congestion avoidance protocols like Transmission Control Protocol (TCP) and to limit the impact of non-adaptive UDP-based applications, the Internet Engineering Task Force has recommended the widespread deployment of active queue management schemes. We describe in this paper a new Random Early Detection (RED) algorithm for congestion control called Dynamic-RED (DRED) from a gradient optimization perspective. One of the goals of RED schemes is to stabilize the queue lengths in routers. However, the current version of RED does not succeed in this goal because the equilibrium queue length strongly depends on the number of active TCP connections. Using a simple optimization technique, DRED randomly discards packets with a load-dependent probability when a buffer in a router gets congested. DRED is also able to stabilize a router queue occupancy at a level independent of the number of active connections over a wide range of load levels. This is done without estimating the number of active TCP connections or flows and without collecting or analyzing state information on individual flows.

A large number of research studies have been done on rate control algorithms. However, many of the proposed algorithms are complex and not scalable to high speeds and large number of connections. In most cases, it is difficult to quantitatively analyze their stability. Our aim in this paper is to provide insight into the design and analysis of a simple, scalable, and provably stable fair rate allocation algorithm. Using discrete-time linear analysis, we analyze quantitatively the performance of the rate control algorithm.

We describe in this paper a new TCP rate control scheme based on a simple recursive algorithm. The idea behind the algorithm is to match the offered network load to the available resources by modifying at an intermediate network element, the receiver's advertised window in TCP acknowledgments returning to the source. We show through simulations that the scheme can efficiently control TCP traffic to limit queue buildups and buffer requirements at the network nodes, resulting in significant improvements in delay, packet loss rates, fairness, throughput and link utilization. Copyright © 2001 John Wiley & Sons, Ltd.

This article describes a technique for weighted proportional window control of elastic traffic such as that generated by TCP. This is achieved through the modification of the receiver's advertised window of TCP connections sharing the bottleneck link while taking into account the price that each user of a connection has paid for the service and the total number of active connections sharing the bottleneck link. Copyright © 2001 John Wiley & Sons, Ltd.

This paper describes fast rate computation (FASTRAC), an explicit rate flow control algorithm for available bit rate (ABR) traffic. Using digital control theory, we develop a simple rate controller for the ABR flow control process. We prove that the controller is stable, fair to all participating sources and configurable with respect to responsiveness. The analysis presented shows that stability of the flow control process depends primarily on two factors, the control update rate and the feedback delay. The implementation of the proposed algorithm is much simpler than other fair rate allocation algorithms. The proposed algorithm demonstrates the ability to scale with speed, distance, different feedback delays, number of users, and number of nodes while remaining robust, efficient, and fair under stressing and dynamic traffic conditions. Copyright © 2001 John Wiley & Sons, Ltd.