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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 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.

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.

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.

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.

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.

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.

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.

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.