Bottleneck Link Research Articles

A pricing approach for bandwidth allocation in differentiated service networks

We present a decentralized auction-based scheme for bandwidth allocation and pricing in a differentiated service-based network. Different classes of clients provide their own expected bandwidth price and required amount of bandwidth. A service provider decides the admission price and differentiated service provided for each class of clients. We introduce mathematical formulations and provide a solution approach to maximize the service provider's revenue. For a single bottleneck link case, a polynomial-time algorithm is developed to obtain an optimal solution. For a multiple bottleneck link case, a heuristic algorithm is developed to solve the problem. The effectiveness of our solution approach is validated by simulation results, which compare our algorithms against the traditional flat-rate pricing approach.

Feedback-Based Adaptive Packet Marking for Proportional Bandwidth Allocation

Differentiated service (DiffServ) networks have been proposed to assure the achievable minimum bandwidth to aggregate flows. However, analyses in the literature show that the current DiffServ networks are biased in favor of ah aggregate flow that has a smaller committed information rate (CIR) when aggregate flows with different CIRs share a bottleneck link. In order to mitigate this unfairness problem, we propose an adaptive marking scheme which provides the relative bandwidth assurance in proportion to the CIRs of the aggregates. By introducing a virtual target rate (VTR) and adjusting it depending on the provision level of the network, each aggregate can obtain its fair share of the bandwidth, regardless of traffic load. This scheme is based on a feedback approach. It utilizes only two-bit feedback information conveyed in the packet header and can be implemented in a distributed manner. Furthermore, the proposed scheme does not require calculating fair shares of aggregates or any additional signaling protocol. Using steady state analysis and extensive simulations, we show that the scheme can provide aggregate flows with their fair shares of bandwidth, which is proportional to the CIRs, under various network conditions

Integrated Proactive Admission Control Technique For both UDP And TCP Traffic Flows

Real time traffic adopting UDP at the transport layer needs some quality of service. It is offered through an admission control scheme. This paper adopts one such scheme which is extended for elastic traffics adopting TCP at the transport layer. The proposed scheme operates on reserving network resources on a proactive manner. It is based on the principle of telephone networks Erlang-B model. The blocking probability measured is used as a flow admission decision parameter. The effectiveness of the proposed admission control algorithm is determined here through simulation. It offers a fair admission rate to both UDP and TCP traffic flows. It also results in a better bottleneck link utilization at a comparatively lower overhead traffic.

TCP Window Control for Variable Bandwidth in Wireless Cellular Networks

Most of TCP schemes in wireless networks assume that the bandwidth of the bottleneck link remains constant over time. However, in wireless cellular networks, to effectively manage the limited resources, the bandwidth is controlled based on radio condition over time. Such varying bandwidth can cause the networks congestion or underutilization. In this letter, we propose a new window control algorithm to improve TCP performance in wireless cellular networks with variable bandwidth. Simulation results illustrate that our proposal improves the performance of TCP in terms of fairness and link utilization

Self-tuning TCP traffic controller using gain margin specification

The paper proposes a self-tuning active queue management (AQM) controller for Internet protocol (IP) routers supporting transmission control protocol (TCP) traffic in the Internet. Classical control theory is applied in the design to meet the frequency-domain-specification gain margin. The user can assign a proper gain margin to achieve good AQM performance. Simple equations for obtaining the AQM controller are derived. The designed AQM controller self-tunes, only when changes in Internet traffic have drifted the traffic monitoring parameter outside its specified interval, otherwise the self-tuning AQM controller remains unchanged. The OPNET® simulations used by the authors in single and in multiple bottleneck links demonstrate that, with the use of a self-tuning AQM controller in the routers, this AQM control system adapts very well to great fluctuations in the TCP traffic, thus providing the network with good stability robustness.

Flow-level QoS for a dynamic load of rate adaptive sessions sharing a bottleneck link

We consider the flow-level quality of service (QoS) seen by a dynamic load of rate adaptive sessions sharing a bottleneck link based on fair share bandwidth allocation. This is of interest both in considering wired networks supporting rate adaptive multimedia sessions and wireless networks supporting voice with rate adaptation to realize graceful degradation during congested periods. Two QoS metrics are considered: the time-average instantaneous utility of the allocated bandwidth, and the time-average of transition penalties associated with the changes in allocation seen by a flow. We present a simple model for rate adaptation, where (heterogeneous) flows can vary their rates within (different) ranges, and present closed-form results for these perceived flow-level QoS metrics. We then prove asymptotic results for large capacity systems exhibiting the salient features of rate adaptation in a dynamic network. Finally, we provide a concrete example, showing how the QoS seen by sessions with different degrees of adaptivity would vary under a natural fair bandwidth allocation policy.

Analytical and experimental evaluation of TCP with an Additive Increase Smooth Decrease (AISD) strategy

This paper presents an analytical model to study the throughput performance of TCP with an Additive Increase Smooth Decrease (AISD) strategy. The new AISD-based TCP protocol produces smooth transfer rates and performs the flow and congestion control functions of TCP. As a result, it is suitable for streaming applications and solves the unfairness problem that occurs when TCP and UDP share the same bottleneck link. The AISD strategy is very simple to implement, only modifies the multiplicative part of the Additive Increase Multiplicative Decrease strategy of TCP, and can be implemented in any TCP version. The smooth part of the strategy is implemented using a low pass filter that considers history in the calculation of the congestion window. The modeling of this new strategy raises new challenges compared to the classical TCP modeling in two ways: first, it needs to be adapted to a more complex dynamism of the congestion window, and second, the model needs to incorporate a scheduler that periodically updates the value of the congestion window. The model is progressively built to finally characterize the steady-state send rate and throughput of a flow as a function of the loss probability, the round-trip time (RTT), the time-out interval, and the scheduler interval. The performance of this AISD-based TCP is compared analytically and experimentally with TCP Reno, and its superior performance is demonstrated.

The probe gap model can underestimate the available bandwidth of multihop paths

The Probe Gap Model (PGM) was proposed as a lightweight and fast available bandwidth estimation method. Measurement tools such as Delphi and Spruce are based on PGM. Compared to estimation methods that require multiple iterations with different probing rates, PGM uses a single probing rate and it infers the available bandwidth from a direct relation between the input and output rates of measurement packet pairs. An important assumption behind the PGM model is that the measured path has a single bottleneck link that determines the available bandwidth of the end-to-end path. In this letter, we show that, even though PGM is accurate in the case of a single queue, it cannot estimate the available bandwidth of multi-hop paths, even if there is a single bottleneck in the path. Whether PGM is accurate or not depends on the routing of cross traffic relative to the measurement traffic. PGM is accurate when the cross traffic follows the same path with the measurement traffic. In the general case, however, PGM can significantly underestimate the available bandwidth of an end-to-end path.

REFWA: An Efficient and Fair Congestion Control Scheme for LEO Satellite Networks

This paper examines some issues that affect the efficiency and fairness of the Transmission Control Protocol (TCP), the backbone of Internet protocol communication, in multi-hops satellite network systems. It proposes a scheme that allows satellite systems to automatically adapt to any change in the number of active TCP flows due to handover occurrence, the free buffer size, and the bandwidth-delay product of the network. The proposed scheme has two major design goals: increasing the system efficiency, and improving its fairness. The system efficiency is controlled by matching the aggregate traffic rate to the sum of the link capacity and total buffer size. On the other hand, the system min-max fairness is achieved by allocating bandwidth among individual flows in proportion with their RTTs. The proposed scheme is dubbed Recursive, Explicit, and Fair Window Adjustment (REFWA). Simulation results elucidate that the REFWA scheme substantially improves the system fairness, reduces the number of packet drops, and makes better utilization of the bottleneck link. The results demonstrate also that the proposed scheme works properly in more complicated environments where connections traverse multiple bottlenecks and the available bandwidth may change over data transmission time

Diagnosis of capacity bottlenecks via passive monitoring in 3G networks: An empirical analysis

In this work we address the problem of inferring the presence of a capacity bottleneck from passive measurements in a 3G mobile network. The study is based on one month of packet traces collected in the UMTS core network of mobilkom austria AG & Co KG, the leading mobile telecommunications provider in Austria, EU. During the measurement period a bottleneck link in the UMTS core network was revealed and removed, therefore the traces enable the accurate analysis and comparison of the traffic behavior in the two network conditions: with and without a capacity bottleneck. Two approaches to bottleneck detection are investigated. The first one is based on the signal analysis of the marginal rate distribution of the traffic aggregate along one day cycle. Since TCP-controlled traffic dominates the overall traffic mix, the presence of a bottleneck strains the aggregate rate distribution and compresses it against the capacity limit during the peak hour. The second approach is based on the analysis of several TCP performance parameters, e.g. estimated frequency of retransmissions. Such statistics are unstable due to the presence of few top users, but this effect can be counteracted with simple filtering methods. Both approaches are validated via simulations. Our results show that both approaches can be used to provide early warning about future occurrences of capacity bottlenecks, and can complement other existing monitoring tools in the operation of a production network.

QuickProbe

An accurate estimate of the available (or unused) bandwidth in a network path can be useful in many applications, including route selection in an overlay or multi-homed network, initial bit-rate selection for video streams or improving the slow start phase of existing TCP protocols. Previously proposed methods for estimating available bandwidth include Pathload [2], PTR [1], Spruce [4], and references therein. The Pathload technique has been shown to be reasonably accurate under a wide range of conditions by independent researchers [3,5]. PTR is more efficient and has been shown to have accuracy similar to Pathload in a more limited set of experiments [1]. With an accurate measure of the bottleneck link capacity, Spruce has been found to be more accurate than Pathload when a new cross traffic stream with known rate is injected. The principal drawback of these techniques is that they require on the order of hundreds of probe packets, and several to several tens of seconds to obtain the available bandwidth estimate. This paper develops a new bandwidth estimation technique, ‘QuickProbe’, that uses 19 probe packets to obtain a conservative estimate of the available bandwidth within a single round trip, and then uses 9-17 further probe packets in each subsequent round trip to refine the estimate. We have compared the QuickProbe and Pathload estimates for hundreds of Internet paths between PlanetLab and other nodes. The paths have measured round-trip times in the range of 20-800 milliseconds, capacities in the range of 0.1 600 Mb/s, and ratios of available bandwidth to capacity in the range of 5-95%. Over such paths, ‘QuickProbe’ obtains conservative available bandwidth estimates after two round trips that are a within a factor of 0.7 1.0 times the Pathload estimate.

Improving the fairness of FAST TCP to new flows

It has been observed that FAST TCP, and the related protocol TCP Vegas, suffer unfairness when many flows arrive at a single bottleneck link, without intervening departures. We show that the effect is even more marked if a new flow arrives when existing flows share bandwidth fairly, and propose a simple method to ameliorate this effect.

Lightweight proactive queue management

The quest for better resource control has been the driving force behind Active Queue Management (AQM) research. Random Early Detection (RED), the defacto standard and its variants have been proposed as simple solutions to the AQM problem. These approaches, however, are known to suffer from problems like parameter sensitivity and inability to capture input traffic load fluctuations accurately, thereby resulting in instability. This paper presents a proactive queue management algorithm called PAQMAN that captures input traffic load fluctuations accurately and regulates the queue size around the desirable level. PAQMAN draws from the predictability in the underlying traffic by employing the Recursive Least Squares (RLS) algorithm to forecast the average queue size over the next prediction interval using the average queue size information of the past intervals. The packet drop probability is then computed as a function of this predicted average queue size. The performance of PAQMAN has been evaluated and compared against existing AQM schemes through ns-2 simulations that encompass varying network conditions for networks comprising of single as well as multiple bottleneck links. Simulation results demonstrate that PAQMAN maintains a relatively low queue size, while at the same time achieving high link utilization and low packet loss. Moreover, the computational overhead of PAQMAN is negligible (lightweight) which further justifies its use.

Experimenting with τ-AIMD over wireless asynchronous networks

The work describes the performance of two congestion control algorithms: AIMD and τ-AIMD. The first is the default mechanism of TCP; the second is a proposed congestion control algorithm that improves fairness of AIMD. We consider asynchronous networks where TCP flows have different propagation delays, a portion of their link is wireless, and they compete for resources over a single bottleneck link. We show that τ-AIMD improves the performance of flows that have long propagation delay and the fairness of the network. In wireless links τ-AIMD outperforms AIMD and the cost of lost packets (or wasted energy) is the same as that of AIMD.

A hybrid active queue management for stability and fast adaptation

The domination of the Internet by TCP-based services has spawned many efforts to provide high network utilization with low loss and delay in a simple and scalable manner. Active queue management (AQM) algorithms attempt to achieve these goals by regulating queues at bottleneck links to provide useful feedback to TCP sources. While many AQM algorithms have been proposed, most suffer from instability, require careful configuration of non-intuitive control parameters, or are not practical because of slow response to dynamic traffic changes. In this paper, we propose a new AQM algorithm, hybrid random early detection (HRED), that combines the more effective elements of recent algorithms with a random early detection (RED) core. HRED maps instantaneous queue length to a drop probability, automatically adjusting the slope and intercept of the mapping function to account for changes in traffic load and to keep queue length within the desired operating range. We demonstrate that straightforward selection of HRED parameters results in stable operation under steady load and rapid adaptation to changes in load. Simulation and implementation tests confirm this stability, and indicate that overall performances of HRED are substantially better than those of earlier AQM algorithms. Finally, HRED control parameters provide several intuitive approaches to trading between required memory, queue stability, and response time.

Proxy MAP for Intra-domain Route Optimization in Hierarchical Mobile IP

The capability of Hierarchical Mobile IP (HMIP) for intra-domain route optimization is impaired when it is combined with Network Mobility (NEMO) technology. Deviations from the optimum path, caused by traffic aggregation in the Mobility Anchor Point (MAP), can be observed within a hierarchical domain. The problem is particularly noticeable in domains that span the mesh network topology. The lack of intra-domain path optimization in multi-level Mobile IP (MIP) leads to inefficient use of network resources. A Proxy Mobility Anchor Point (PMAP) functionality is proposed in domain nodes to enable intra-domain path optimization in multi-level MIP. Numerical evaluation and simulations indicate that this proposal can improve routing efficiency and throughput. The solution can be especially rewarding in network architectures where access network is separated from global network by bottleneck links and where the majority of users accessing the network are mobile routers.

Public review for LTCP

LTCP is a new modification to TCP's congestion control to enable it to scale to high-speed networks. In comparison with prior work on high-speed congestion control, the paper explores an interesting design space: what if a single flow acts similarly to K virtual TCPs, where K is dynamically adjusted according to the network's congestion state; K increases in times of under-load to aggressively grab the available bandwidth and decreases in times of congestion to quickly release the bandwidth. The paper elaborates on how to design such a protocol and particularly how to make it fair when multiple flows with a different initial K share the same bottleneckThough I find it interesting to explore such a design, with much prior work on high-speed congestion control, it is unclear what LTCP can add. The paper claims that one of the contributions of LTCP is that "it retains the time-tested AIMD behavior of TCP." Yet, because of the dynamic value of K, LTCP does not really do AIMD. Besides, why is maintaining AIMD important? The paper does not elaborate on this issue; neither does it show that the use of different increase-decrease rule has hurt other high-speed congestion control protocolsOverall, LTCP is a nice piece of research but I would encourage the authors to argue more why their design is important given all prior work on the topic. Further, they should run more extensive simulations that explore multiple bottleneck links, and provide at least a simple analysis that supports the convergence claim.

BUFFER EFFICIENCY IN WORMHOLE ROUTERS

AbstractThe efficiency of buffer usage in wormhole torus networks is evaluated withemphasis on new routing algorithms. These algorithms use hardwired dateline and allow flits of different packets to co-reside in buffers along escape virtual channels so that the bottleneck links and bubble problems in existing algorithms can be alleviated. The algorithms are demonstrated to be deadlock-free and more efficient in use of buffer space than their counterparts.

Buffer Efficiency In Wormhole Routers

The efficiency of buffer usage in wormhole torus networks is evaluated withemphasis on new routing algorithms. These algorithms use hardwired dateline and allow flits of different packets to co-reside in buffers along escape virtual channels so that the bottleneck links and bubble problems in existing algorithms can be alleviated. The algorithms are demonstrated to be deadlock-free and more efficient in use of buffer space than their counterparts.

Traffic characteristics of aggregated module downloads for mobile terminal reconfiguration

In recent years, there has been an increasing proliferation of reasons for software/firmware in desktop computers to be upgraded over the Internet, and as computational devices continue to become ever more advanced and compact, it is anticipated that the same will be the case for mobile terminals in the near future. In this paper, we consider some important characteristics of such upgrade traffic. Initially, we study file size distributions for various file types on a range of servers/systems, and conclude that the Pareto distribution, as is often used to represent file sizes, is inappropriate. In fact, in order to achieve a better fit to the bulk of the distribution while still maintaining an infinite variance, file sizes are usually far better represented by the Pareto type 2 (Lomax) distribution. Leading on from this, we simulate multiple concurrent files being downloaded over a bottleneck link, taken from particular file-size distributions with a focus on mobile terminal reconfiguration. In doing so, we investigate the degree of burstiness/self-similarity and correlation structure in the resulting traffic. Understanding these phenomena is particularly important because of their tendencies to increase packet losses and queuing delays through compound fluctuations in link loads.