Bandwidth-delay Product Research Articles

QRM: A queue rate management for fairness and TCP flooding protection in mission-critical networks

When statistical multiplexing is used to provide connectivity to a number of client hosts through a high-delay link, the original TCP as well as TCP variants designed to improve performance on those links often provide poor performance and sub-optimal QoS properties. Centralised and collaborative resource management tools like C2MLhave been proposed to guarantee intra-protocol fairness, inter-protocol friendliness, low queues utilisation and optimal throughput along with the reliable delivery of packets. However, such tools offer only very limited security guarantees. Both good citizenship and security from flooding attacks are fundamental conditions for the provision of fairness, especially in mission-critical networks. For example, perpetrators of a man-provoked disaster may want to perform a resource exhaustion attack on the network supporting disaster recovery operations, so as to cut out legitimate users from the communications and increase the emergency impact. In this paper we present Queue Rate Management (QRM), an Active Queue Management scheme able to provide protection from traffic overflow attacks in scenarios where access to the shared link is controlled by a tool that assigns to client hosts a bandwidth upper bound. The proposed algorithm checks whether a node is exceeding its allowed rate, and consequently decides whether to keep or drop packets coming from that host. We mathematically prove that with QRMthe gateway queue size can never exceed the Bandwidth-Delay Product of the channel. Furthermore, we employ the ns-3 network simulator to compare QRMwith CoDel, RED and GREEN, showing how QRMprovides better performance in terms of both throughput and QoS guarantees in the aforementioned scenarios.

HBDP 네트워크에서 대역폭 점유와 RTT 공정성 향상을 위한 네트워크 적응적 혼잡제어 기법

These days, the networks have exhibited HBDP (High Bandwidth Delay Product) characteristics. The legacy TCP slowly increases the size of the congestion window and drastically decreases the size of a congestion window. The legacy TCP has been found to be unsuitable for HBDP networks. TCP mechanisms for solving the problems of the legacy TCP can be categorized into the loss-based TCP and the delay-based TCP. Most of the TCP mechanisms use the standard slow start phase, which leads to a heavy packet loss event caused by the overshoot. Also, in the case of congestion avoidance, the loss-based TCP has shown problems of wastage in terms of the bandwidth and RTT (Round Trip Time) fairness. The delay-based TCP has shown a slow increase in speed and low occupancy of the bandwidth. In this paper, we propose a new scheme for improving the over shoot, increasing the speed of the bandwidth and overcoming the bandwidth occupancy and RTT fairness issues. By monitoring the buffer condition in the bottleneck link, the proposed scheme does congestion control and solves problems of slow start and congestion avoidance. By evaluating performance, we prove that our proposed scheme offers better performance in HBDP networks compared to the previous TCP mechanisms.

Slow-light plasmonic metamaterial based on dressed-state analog of electromagnetically induced transparency.

We consider a simple configuration for realizing one-dimensional slow-light metamaterials with large bandwidth-delay products using stub-shaped Fabry-Perot resonators as building blocks. Each meta-atom gives rise to large group indices because of a classical analog of the dressed-state picture of electromagnetically induced transparency. By connecting up to eight meta-atoms, we find bandwidth-delay products over unity and group indices approaching 100. Our approach is quite general and can be applied to any type of Fabry-Perot resonators and tuned to different operating wavelengths.

Evolution of TCP in High Speed Networks

TCP congestion control protocols have low performances in high speed wide area networks mainly because their slow response with large congestion windows. This TCP behavior has initiated new design phase of alternative protocols that provide improved traffic utilization in high bandwidth delay product networks. The paper presents survey of various high speed sender side congestion control proposals that preserve the fundamental host to host principle. Solutions focus on variety of problems occurring in high speed environment with intention to eliminate congestion collapses and to ensure effective resource utilization. Internet data transfer does not depend only on that how TCP will utilize the network capacity, we have to stress that TCP must cooperate with existing transmitting data protocols through the same network in order to assure fair resource sharing. Part of the paper scope are state of the art high speed TCP proposals, we explore their congestion control techniques, strengths, weaknesses and we try to detect future TCP development possibilities.

EBDP BUFFER SIZING STRATEGY 802.11 BASED WLANS

In this paper we present wired routers, for whom the sizing of buffers is an active research topic. The classical rule of thumb for sizing wired buffers is to set buffer sizes to be the product of the bandwidth and the average delay of the flows utilizing this link, namely the Bandwidth-Delay Product (BDP) rule. Surprisingly, however the sizing of buffers in wireless networks (especially those based on 802.11/802.11e) appears to have received very little attention within the networking community. Exceptions include the recent work in relating to buffer sizing for voice traffic in 802.11e WLANs, work in which considers the impact of buffer sizing on TCP upload/download fairness, and work in which is related to 802.11e parameter settings.

Active queue management with discrete sliding modes in TCP networks

Abstract In this paper, a new active queue management (AQM) algorithm for data traffic control in TCP/IP networks is developed. The algorithm design is based on the principles of discrete sliding-mode control. Unlike majority of earlier studies, the design procedure considers the effects of both non-negligible delay in transferring data and feedback information and unpredictable capacity variations. The switching function is selected to incorporate a delay compensation mechanism, which ensures efficient network operation even for large bandwidthdelay product connections. The proposed algorithm, implemented as a packet marking scheme, is tested in discrete event ns-2 simulator. The results show that the algorithm provides fast convergence to steady state after sudden, unanticipated capacity changes. By generating smaller overshoots, the proposed algorithm also allows for reducing buffer space requirements to avoid packet loss as compared to the benchmark AQM solutions.

Enhanced TFRC for high quality video streaming over high bandwidth delay product networks

Transmission control protocol friendly rate control (TFRC) is designed to mainly provide optimal service for unicast applications, such as multimedia streaming in the best-effort Internet environment. However, high bandwidth networks with large delays present an environment where TFRC may have a problem in utilizing the full bandwidth. TFRC inherits the slow-start mechanism of TCP Reno, but this is a time-consuming process that may require many round-trip-times (RTTs), until an appropriate sending rate is reached. Another disadvantage inherited from TCP Reno is the RTT-unfairness problem, which severely affects the performance of long-RTT flows. In this paper, we suggest enhanced TFRC for high quality video streaming over high bandwidth delay product networks. First, we propose a fast startup scheme that increases the data rate more aggressively than the slow-start, while mitigating the overshooting problem. Second, we propose a bandwidth estimation method to achieve more equitable bandwidth allocations among streaming flows that compete for the same narrow link with different RTTs. Finally, we improve the responsiveness of TFRC in the presence of severe congestion. Simulation results have shown that our proposal can achieve a fast startup and provide fairness with competing flows compared to the original TFRC.

Comparative study of high-speed Linux TCP variants over high-BDP networks

Transmission Control Protocol (TCP) has been profusely used by most of internet applications. Since 1970s, several TCP variants have been developed in order to cope with the fast increasing of network capacities especially in high Bandwidth Delay Product (high-BDP) networks. In these TCP variants, several approaches have been used, some of these approaches have the ability to estimate available bandwidths and some react based on network loss and/or delay changes. This variety of the used approaches arises many consequent problems with different levels of dependability and accuracy. Indeed, a particular TCP variant which is proper for wireless networks, may not fit for high-BDP wired networks and vice versa. Therefore, it is necessary to conduct a comparison between the high-speed TCP variants that have a high level of importance especially after the fast growth of networks bandwidths. In this paper, high-speed TCP variants, that are implemented in Linux and available for research, have been evaluated using NS2 network simulator. This performance evaluation presents the advantages and disadvantages of these TCP variants in terms of throughput, loss-ratio and fairness over high-BDP networks. The results reveal that, CUBIC and YeAH overcome the other high-speed TCP variants in different cases of buffer size. However, they still require more improvement to extend their ability to fully utilize the high-speed bandwidths, especially when the applied buffer is near-zero or less than the BDP of the link.

Adaptive congestion control framework and a simple implementation on high bandwidth-delay product networks

As new link technologies and sub-networks proliferate and evolve, a large number of TCP variants have been developed for different types of the network environments. They can lead to major performance gains by taking advantage of local characteristics of the specific network. However, these TCP variants could not be automatically chosen according to the lower network environments. In this paper, we propose the ACCF, an adaptive congestion control framework, which can automatically transition among existing congestion control mechanisms according to the change of the network status. Then we perform a simple implementation of ACCF over the networks with high bandwidth-delay product (BDP). It can switch the congestion control approaches between the delay-based ones and the loss-based ones according to the network status. Extensive experiments are conducted based on network simulators as well as over real wired networks on different time periods of the day. For the simulation measures, the experimental results show that the performance of ACCF is significantly improved as compared to other state-of-the-art algorithms in term of throughput, fairness and TCP-friendliness. For the real network tests, the experimental results show that ACCF achieves speedup ratios up to 225.83% compared with average throughput of other TCP congestion control algorithms.

Sharing Bandwidth by Allocating Switch Buffer in Data Center Networks

In today's data centers, the round trip propagation delay is quite small. Therefore, switch buffer sizes are much larger than the Bandwidth Delay Product (BDP). Based on this observation, in this paper we introduce a new transport protocol which provides bandwidth Sharing by Allocating switch Buffer (SAB) for data centers. SAB sets the congestion windows for flows based on the buffer size of the switches along the path. On one hand, as long as the total buffer allocated to all the flows is larger than the BDP, the network bandwidth can be fully utilized. On the other hand, since SAB only allocates the buffer space to flows, the totally injected traffic will not exceed the network capacity. Thus, SAB rarely loses packets. SAB also reduces flow completion time by allowing flows to reach their fair share of bandwidth quickly. The results of a series of experiments and simulations demonstrate that SAB has the features of fast convergence and rare packet loss. It reduces the latency of short flows and solves theTCP Incast and TCP Outcast problems.

Achieve Significant Throughput Gains in Wireless Networks with Large Delay-Bandwidth Product

Traditionally, Bandwidth-Delay Product can be used to measure the capacity of network “pipe” between two nodes. However, in multi-hop wireless networks, Bandwidth-Delay Product cannot reveal the network condition accurately. In this paper, we define a new metric called Delay-Bandwidth Product (DBP) for wireless networks, which measures the capacity of a one-hop pipe in wireless networks. Wireless networks with a large DBP can have a throughput larger than the one based on traditional understanding. We propose a scheduling algorithm aims for making use of the large DBP in wireless networks. We then design simulations to figure out how much throughput gains can be achieved in wireless networks, with small DBPs and large DBPs respectively. The simulation result demonstrates that we can achieve significant throughput gains in wireless networks with large DBP.

POTENTIAL PERFORMANCE OF THE COMMUNICATION SYSTEMS USING AUTOCORRELATION RECEPTION OF SHIFT-KEYED NOISE SIGNALS

Mathematical models of the signals peculiar to communications using the autocorrelation reception of the shift-keyed noise signals are suggested in the present paper. System performance is investigated and the particularities of the bit error probability dependence upon the normalized signal-to-noise ratio and the bandwidth-delay product are revealed. The definition of the optimal value of the signal-to-noise ratio upon power is substantiated and the relevant values are calculated. The performed comparative analysis proved that the communication using the differential noise shift-keying, within which total time delay diversity is applied between the reference and the data signals, possesses the best system performance among the communications considered in this paper.

SCTP performance improvement based on virtual receiver window

This article puts forward one algorithm for stream control transfer protocol (SCTP) improvement with limited receiver buffer (RBUF). As is well known, SCTP is one of the most important transfer control protocol, but most researches focus on the situations without the RBUF limit. In this study, we analyze the impact of the RBUF size on the performance. Computer simulations show that the network utility is low in reliable transfer, when the RBUF size is smaller than bandwidth delay product. By studying the transmission sequence number (TSN) transfer progress, we find that the peer receiver window (PEER_RWND), which lags behind the true receiver window (RWND), leads to the poor network utility. To improve SCTP performance with limited RBUF, the virtual receiver window (VIRTUAL_RWND) is introduced. Based on the VIRTUAL_RWND, one algorithm is proposed to increase the sending rate. Computer simulations have evaluated an excellent performance of the proposed algorithm at both ideal link without lost packet and nonideal link with lost packet.

Adaptive explicit congestion control based on bandwidth estimation for high bandwidth-delay product networks

Existing congestion control protocols for high bandwidth-delay product (HBDP) networks either suffer from efficiency degradation caused by insufficient feedback of the network state, or are difficult to deploy in real networks because they require modification in IP header to provide enough feedback information. Based on VCP, this paper proposes the VCP-BE protocol that can provide more accurate network state information by estimating available bandwidth for sender with an adaptive bandwidth estimator. The estimator can dynamically adjust its parameters (e.g., observing interval) to achieve different accuracy and responsiveness to adapt to the network state indicated by two ECN bits, without no more modification in IP header than VCP. Furthermore, when the network load is high, VCP-BE calculates the ratio of current throughput to the available bandwidth to judge if network is close to the overload state. With these fine-grained network state estimation, VCP-BE can adjust the congestion window more precisely than VCP, thus greatly improves its convergence speed of achieving high bandwidth utilization and fairness. Simulation results show that VCP-BE also outperforms MLCP, which uses seven bits for explicit feedback while VCP-BE only uses two ECN bits.

Performance Measurement of TCP and UDP Using Different Queuing Algorithm in High Speed Local Area Network

In the current high speed networks, it is increasingly important to have mechanisms that keep throughput high but average queue sizes low. The queue management algorithm, which is applied to a router, plays an important role in providing Quality of Service (QoS). This paper present a Simulation-based performance Evaluation of two well known transport layer protocol TCP and UDP using two popular queue management methods; Random Early Detection (RED) and Drop Tail, in terms of throughput, queuing delay, packet drop rate and bandwidth utilization. In this paper we study the effect of buffer behavior on each one of these QoS measurements. TCP, UDP and shared topology scenarios are simulated by considering varying number of client case topologies and also the effect of performance in TCP and UDP with increasing the number of client. This simulation report is useful for implementing the queue management algorithm in router based on the traffic type and bandwidth. Index Terms—TCP, UDP, drop tail, RED queue, bandwidth delay product, throughput and end-to-end delay.

An Active Queue Management for High Bandwidth-Delay Product Networks

An Active Queue Management for High Bandwidth-Delay Product Networks

Congestion Control With Multipacket Feedback

Many congestion control protocols use explicit feedback from the network to achieve high performance. Most of these either require more bits for feedback than are available in the IP header or incur performance limitations due to inaccurate congestion feedback. There has been recent interest in protocols that obtain high-resolution estimates of congestion by combining the explicit congestion notification (ECN) marks of multiple packets, and using this to guide multiplicative increase, additive increase, multiplicative decrease (MI-AI-MD) window adaptation. This paper studies the potential of such approaches, both analytically and by simulation. The evaluation focuses on a new protocol called Binary Marking Congestion Control (BMCC). It is shown that these schemes can quickly acquire unused capacity, quickly approach a fair rate distribution, and have relatively smooth sending rates, even on high bandwidth-delay product networks. This is achieved while maintaining low average queue length and negligible packet loss. Using extensive simulations, we show that BMCC outperforms XCP, VCP, MLCP, CUBIC, CTCP, SACK, and in some cases RCP, in terms of average flow completion times. Suggestions are also given for the incremental deployment of BMCC.

Efficient and low-delay error control for large-BDP networks

Today’s dominant Internet transport protocol – TCP – is known to achieve a low bandwidth utilization on lossy large bandwidth-delay product paths. As such paths become common in the Internet, TCP’s inefficiency will be the main bottleneck in deploying high-rate, low-latency communications services such as tele-presence, free-viewpoint video or virtual reality. In this paper, we present PRRT –Predictably Reliable Real-time Transport, a protocol layer that efficiently supports these services. PRRT is based on a hybrid error correction that combines proactive and reactive reliability and dynamically adjusts the amount of redundancy to path properties. By relying on an analytical model of the path, it achieves a near-optimal packet-erasure correction under service-specific end-to-end delay. As a result, it simultaneously achieves a low latency and a high bandwidth utilization. We describe key features of the PRRT: (1) adaptive block-erasure coding and (2) predictable reliability based on a statistical performance model. We implement PRRT in the Linux kernel and evaluate its performance over a range of large bandwidth-delay product paths under a strict service-specific delay.

Asymptotic Stability Analysis of E-speed Start Congestion Control Protocol for TCP

All the proposed window based congestion control protocols use a single slow start algorithm. It has been shown in literature that slow start inefficiently utilizes bandwidth in a network with high bandwidth delay product (BDP). A multiple startup algorithm (E- speed start) was proposed in this work. E-speed start specifies a multiple selectable startup for Transmission Control Protocol (TCP). It is proposed that startup speed for TCP be selectable from n-arry startup algorithms. In E-speed start, a binary start was implemented based on a calculated value β which depends on the available bandwidth and other network conditions. A model for E- speed start was derived using utility maximization theory and the principle of decentralised solution. The model derived was shown to be stable by linear approximation, Laplace transform and by using the concept of Nyquist Stability criterion.

Periodic early detection for improved TCP performance and energy efficiency

Reducing the size of packet buffers in network equipment is a straightforward method for improving the network performance experienced by user applications and also the energy efficiency of system designs. Smaller buffers imply lower queueing delays, with faster delivery of data to receivers and shorter round-trip times for better controlling the size of TCP congestion windows. If small enough, downsized buffers can even fit in the same chips where packets are processed and scheduled, avoiding the energy cost of external memory chips and of the interfaces that drive them. On-chip buffer memories also abate packet access latencies, further contributing to system scalability and bandwidth density. Unfortunately, despite more than two decades of intense research activity on buffer management, current-day system designs still rely on the conventional bandwidth-delay product rule to set the size of their buffers. Instead of decreasing, buffer sizes keep on growing linearly with link capacities.We draw from the limitations of the buffer management schemes that are commonly available in commercial network equipment to define Periodic Early Detection (PED), a new active queue management scheme that achieves important buffer size reductions (more than 95%) while retaining TCP throughput and fairness. We show that PED enables on-chip buffer implementations for link rates up to 100Gbps while relieving end users from network performance disruptions of common occurrence.