Detecting Fair Queuing for Better Congestion Control

Total Page:16

File Type:pdf, Size:1020Kb

Detecting Fair Queuing for Better Congestion Control Detecting Fair Queuing for Better Congestion Control Maximilian Bachl, Joachim Fabini, Tanja Zseby Technische Universität Wien fi[email protected] Abstract—Low delay is an explicit requirement for applications While FQ solves many problems regarding Congestion such as cloud gaming and video conferencing. Delay-based con- Control (CC), it is still not ubiquitously deployed. Flows can gestion control can achieve the same throughput but significantly benefit from knowledge of FQ enabled bottleneck links on the smaller delay than loss-based one and is thus ideal for these applications. However, when a delay- and a loss-based flow path to dynamically adapt their CC. In the case of FQ they can compete for a bottleneck, the loss-based one can monopolize all use a delay-based CCA and otherwise revert to a loss-based the bandwidth and starve the delay-based one. Fair queuing at the one. Our contribution is the design and evaluation of such bottleneck link solves this problem by assigning an equal share of a mechanism that determines the presence of FQ during the the available bandwidth to each flow. However, so far no end host startup phase of a flow’s CCA and sets the CCA accordingly based algorithm to detect fair queuing exists. Our contribution is the development of an algorithm that detects fair queuing at flow at the end of the startup. startup and chooses delay-based congestion control if there is fair We show that this solution reliably determines the presence queuing. Otherwise, loss-based congestion control can be used as of fair queuing and that by using this approach, queuing delay a backup option. Results show that our algorithm reliably detects can be considerably lowered while maintaining throughput fair queuing and can achieve low delay and high throughput in high if FQ is detected. In case no FQ is enabled, our algorithm case fair queuing is detected. detects this as well and a loss-based CC is used, which does not starve when facing competition from other loss-based I. INTRODUCTION flows. While our mechanism is beneficial for network flows in Emerging applications such as cloud gaming [1] and remote general, we argue that it is most useful for long-running delay- virtual reality [2] applications require high throughput and low sensitive flows, such as cloud gaming, video conferencing etc. delay. Furthermore, ultra low latency communications have To make our work reproducible and to encourage further been made a priority for 5G [3]. We argue that for achieving research, we make the code, the results and the figures publicly high throughput as well as low delay, congestion control must available1. be taken into account. Delay-based Congestion Control Algorithms (CCAs) were II. RELATED WORK proposed to provide high throughput and low delay for net- Our work depends on active measurements for the startup work flows. They use the queuing delay as an indication of phase of CC and proposes a new measurement technique congestion and lower their throughput as soon as the delay to detect Active Queue Management (AQM). This section increases. Such approaches have been proposed in the last discusses preliminary work related to these fields. century [4] and recently have seen a surge in popularity [5, 6, 7, 8]. While delay-based CCAs fulfill their goal of A. CC using Active Measurements low delay and high throughput, they cannot handle competing Several approaches have been recently proposed which flows with different CCAs well [9, 10]. This is especially aim to incorporate active measurements into CCAs. [15, 16] arXiv:2010.08362v3 [cs.NI] 19 Feb 2021 prevalent when they compete against loss-based CCAs, the introduced the CCA PCC, which uses active measurements to latter being more “aggressive”. While the delay-based CCAs find the sending rate which optimizes a given reward function. back off as soon as queuing delay increases, the loss-based [8] introduce the BBR CCA, which uses active measurement ones continue increasing their throughput until the buffer is to create a model of the network connection. This model is full and packet loss occurs. This results in the loss-based flows then used to determine the optimal sending rate. An approach monopolize the available bandwidth and starve the delay-based proposed by [17] aims to differentiate bulk transfer flows, ones [11, 12, 13]. which they deem “elastic” because they try to grab as much This unfairness can be mitigated by using Fair Queuing bandwidth as available, from other flows that send at a constant (FQ) at the bottleneck link, isolating each flow from all other rate, which they call “unelastic”. They use this elasticity flows and assigning each flow an equal share of bandwidth detection to determine if delay-based CC can be used or not. If [14]. Consequently, loss-based flows can no longer acquire cross traffic is deemed elastic then they use classic loss-based bandwidth that delay-based flows have released due to their CC and otherwise delay-based CC. network-friendly behavior. In this case, delay-based CCAs achieve their goal of high bandwidth and low delay. 1https://github.com/CN-TU/PCC-Uspace 1 B. Flow startup techniques sending rate receiving rate receiving rate Besides the classical TCP slow start algorithm [18] some ow 2 ow 2 ow 2 other proposals were made such as Quick Start [19], which ow 1 ow 1 ow 1 aims to speed up flow start by routers informing the end point, which rate they deem appropriate. [20] investigate the impact link speed link speed of AQM and CC during flow startup. They inspect a variety of different AQM algorithms and also investigate FQ. [21] propose “chirping” for flow startup to estimate the available bandwidth. This works by sending packets with different gaps time time time between them to verify what the actual link speed is. [22] 1 r 2 rs 3 rs 1 r 2 rs 3 rs 1 r 2 rs 3 rs propose speeding up flow startup by using packets of different (a) Sending rate (b) Receiving rate (no FQ). (c) Receiving rate (FQ). priority. The link is flooded with low-priority packets and Fig. 1: An example illustrating our proposed flow startup in case the link is not fully utilized during startup, these mechanism. Figure 1a shows the sending rate, 1b the receiving low-priority packets are going to be transmitted, allowing the rate in case there’s no FQ and 1c the receiving rate if there is starting flow to achieve high throughput. If the link is already FQ. saturated, the low-priority packets are going to be discarded and no other flow suffers. C. Detecting AQM mechanisms 2) After every Round Trip Time (RTT), if no packet loss occurred, double the sending rate of both flows. There are few publications that investigate methods to detect 3) If packet loss occurred in both flows in the previous RTT, the presence of an AQM. [23] propose a method to detect and calculate the following metric: distinguish certain popular AQM mechanisms. However, un- like our work, it doesn’t consider FQ. [24] propose a machine receiving rate of flow 1 sending rate of flow 1 learning based approach to fingerprint AQM algorithm. FQ is loss ratio = (1) not considered, too. receiving rate of flow 2 sending rate of flow 2 D. Detecting the presence of a shared bottleneck This loss ratio metric indicates if flow 2, which has Several approaches have been proposed to detect whether a higher sending rate, achieves a higher receiving rate in a set of flows, some of them share a common bottleneck (goodput) than flow 1. If this is the case, there’s no FQ, [25, 26, 27]. These approaches typically operate by compar- otherwise there is. ing some statistics of flows to check whether they correlate It is necessary to wait for packet loss in both connections between flows. For example, it can be checked if throughput because only then it is certain that the link is saturated. correlates between two flows. If throughput correlates, it can Our algorithm assumes that the link is saturated. mean that these flows share a common bottleneck link, and if 4) If FQ is used, the loss ratio is 2, otherwise it is 1. We additional flows join at the bottleneck link, all other flows’ choose 1.5 as the cutoff value. throughputs go down, which is the reason why they are 5) Since the presence or absence of FQ is now known the correlated. While approaches to detect a shared bottleneck link appropriate CCA can be launched. appear to be similar to what this paper is aiming to do, there are differences: Two flows can share a common bottleneck link Figure 1a, 1b and 1c schematically illustrate this mecha- but the bottleneck link can have fair queuing. Thus the aims nism. We assume a sender being connected to a receiver, with of shared bottleneck and fair queuing detection are different. a bottleneck link between them. We also assume that all flows between the source and the destination follow the same route. III. CONCEPT On this bottleneck link, the queue is either managed by FQ The overall concept is the following: or by a shared queue (no FQ). 1) During flow startup, determine whether the bottleneck Figure 1a shows how the sending rate (observed before the link deploys FQ or not. bottleneck link) is doubled after each RTT. Furthermore, it 2) If FQ is used change to a delay-based CCA or revert to shows that flow 2 has double the sending rate of flow 1. a loss-based CCA otherwise. Figure 1b shows the receiving rate at the receiver after the bottleneck link, if this bottleneck link has a a shared queue A.
Recommended publications
  • A Comparison of Mechanisms for Improving TCP Performance Over Wireless Links
    A Comparison of Mechanisms for Improving TCP Performance over Wireless Links Hari Balakrishnan, Venkata N. Padmanabhan, Srinivasan Seshan and Randy H. Katz1 {hari,padmanab,ss,randy}@cs.berkeley.edu Computer Science Division, Department of EECS, University of California at Berkeley Abstract the estimated round-trip delay and the mean linear deviation from it. The sender identifies the loss of a packet either by Reliable transport protocols such as TCP are tuned to per- the arrival of several duplicate cumulative acknowledg- form well in traditional networks where packet losses occur ments or the absence of an acknowledgment for the packet mostly because of congestion. However, networks with within a timeout interval equal to the sum of the smoothed wireless and other lossy links also suffer from significant round-trip delay and four times its mean deviation. TCP losses due to bit errors and handoffs. TCP responds to all reacts to packet losses by dropping its transmission (conges- losses by invoking congestion control and avoidance algo- tion) window size before retransmitting packets, initiating rithms, resulting in degraded end-to-end performance in congestion control or avoidance mechanisms (e.g., slow wireless and lossy systems. In this paper, we compare sev- start [13]) and backing off its retransmission timer (Karn’s eral schemes designed to improve the performance of TCP Algorithm [16]). These measures result in a reduction in the in such networks. We classify these schemes into three load on the intermediate links, thereby controlling the con- broad categories: end-to-end protocols, where loss recovery gestion in the network. is performed by the sender; link-layer protocols, that pro- vide local reliability; and split-connection protocols, that Unfortunately, when packets are lost in networks for rea- break the end-to-end connection into two parts at the base sons other than congestion, these measures result in an station.
    [Show full text]
  • Dynamic Optimization of the Quality of Experience During Mobile Video Watching
    View metadata, citation and similar papers at core.ac.uk brought to you by CORE provided by Ghent University Academic Bibliography Dynamic Optimization of the Quality of Experience during Mobile Video Watching Toon De Pessemier, Luc Martens, Wout Joseph iMinds - Ghent University, Dept. of Information Technology G. Crommenlaan 8 box 201, B-9050 Ghent, Belgium Tel: +32 9 33 14908, Fax:+32 9 33 14899 Email: {toon.depessemier, luc.martens, wout.joseph}@intec.ugent.be Abstract—Mobile video consumption through streaming is Traditionally, network operators and service providers used becoming increasingly popular. The video parameters for an to pay close attention to the Quality of Service (QoS), where optimal quality are often automatically determined based on the emphasis was on delivering a fluent service. QoS is defined device and network conditions. Current mobile video services by the ITU-T as “the collective effect of service perfor- typically decide on these parameters before starting the video mance” [3]. The QoS is generally assessed by objectively- streaming and stick to these parameters during video playback. measured video quality metrics. These quality metrics play a However in a mobile environment, conditions may change sig- nificantly during video playback. Therefore, this paper proposes crucial role in meeting the promised QoS and in improving a dynamic optimization of the quality taking into account real- the obtained video quality at the receiver side [4]. time data regarding network, device, and user movement during Although useful, these objective quality metrics only ad- video playback. The optimization method is able to change the dress the perceived quality of a video session partly, since these video quality level during playback if changing conditions require this.
    [Show full text]
  • TCP Optimization Improve Qoe by Managing TCP-Caused Latency
    SOLUTION BRIEF TCP Optimization Improve QoE by managing TCP-caused latency TCP OPTIMIZATION MARKET OVERVIEW The Transmission Control Protocol (TCP) is the engine of the internet and its dominant DELIVERS: role in traffic delivery has only grown as customers demand support for an ever- increasing array of personal, social, and business applications and services. However, Unique Approach TCP is not without its faults, and its well-known weaknesses have created a significant Creates a TCP “midpoint” that takes control of drag on network performance and customer quality of experience (QoE). the TCP connection while remaining end-to-end transparent Given that TCP traffic represents ~90% on fixed and 90%+ on mobile networks, many operators that have overlooked TCP’s shortcomings in the past are now considering new ways Buffer Mangement to manage TCP traffic to increase its performance, service quality, and network utilization. Manages “starving” and “bufferbloats” efficiently by adjusting the sending rate to The very characteristics of TCP that made it very popular and successful led to its performance correspond to the level of buffer data in the challenges in modern day networks. TCP is largely focused on providing fail-safe traffic delivery, access network but this reliability comes at a cost: lower performance, subpar customer experience, and Better Congestion Control underutilized network assets. Operators can delay other congestion management investments TCP is hobbled by its lack of end-to-end network visibility. Without the ability
    [Show full text]
  • An Active-Passive Measurement Study of TCP Performance Over LTE on High-Speed Rails∗
    An Active-Passive Measurement Study of TCP Performance over LTE on High-speed Rails∗ Jing Wang†, Yufan Zheng†, Yunzhe Ni†, Chenren Xu†, Feng Qian], Wangyang Li†, Wantong Jiang† Yihua Cheng†, Zhuo Cheng†, Yuanjie Li§, Xiufeng Xie§, Yi Sun[, Zhongfeng Wang\† †Peking University, China ]University of Minnesota – Twin Cities, USA §Hewlett Packard Labs, USA [Institute of Computing Technology, University of Chinese Academy of Sciences \China Academy of Railway Sciences ABSTRACT CCS CONCEPTS High-speed rail (HSR) systems potentially provide a more • Networks → Network measurement; efficient way of door-to-door transportation than airplane. However, they also pose unprecedented challenges in deliver- KEYWORDS ing seamless Internet service for on-board passengers. In this Measurement, High-speed Rails, TCP, LTE, High Mobility, paper, we conduct a large-scale active-passive measurement Handover, CUBIC, BBR study of TCP performance over LTE on HSR. Our measure- ment targets the HSR routes in China operating at above 300 1 INTRODUCTION km/h. We performed extensive data collection through both Recently, the rapid development of high-speed rails (HSRs) controlled setting and passive monitoring, obtaining 1732.9 has dramatically changed the way people commute for medium- GB data collected over 135719 km of trips. Leveraging such a to-long distance travel. For instance, a train traveling above unique dataset, we measure important performance metrics 300 km/h potentially provides a more efficient way of door- such as TCP goodput, latency, loss rate, as well as key char- to-door transportation than airplane. To date, 18 countries acteristics of TCP flows, application breakdown, and users’ have developed HSR to connect major cities.
    [Show full text]
  • Overall Design of Satellite Networks for Internet Services with Qos Support
    electronics Article Overall Design of Satellite Networks for Internet Services with QoS Support Kyu-Hwan Lee and Kyoung Youl Park * Agency for Defense Development, DaeJeon 43186, Korea; [email protected] * Correspondence: [email protected] Received: 28 May 2019; Accepted: 13 June 2019; Published: 17 June 2019 Abstract: The satellite network is useful in various applications because of its coverage, broadcast capability, costs independent of the distance, and easy deployment. Recently, thanks to the advanced technologies in the satellite communication, the high throughput satellite system with mesh connections has been applied to the Internet backbone. In this paper, we propose a practical overall design of the satellite network to provide Internet services with quality of service (QoS) support via the satellite network. In the proposed design, we consider two service types such as delay-tolerance and delay-sensitive services allowing the long propagation delay of the satellite link. Since it is crucial to evaluate the user satisfaction in the application layer for various environments to provide the QoS support, we also define the performance metrics for the user satisfaction and derive the major factors to be considered in the QoS policy. The results of the performance evaluation show that there are factors such as the traffic load and burstiness in the QoS policy for the delay-tolerance service with volume-based dynamic capacity in the satellite network. For the delay-sensitive service with rate-based dynamic capacity, it is additionally indicated that it is important for the estimation of effective data transmission rate to guarantee the QoS. Furthermore, it is shown that the small data size has an effect on the slight reduction of the QoS performance in the satellite network.
    [Show full text]
  • Wifox: Scaling Wifi Performance for Large Audience Environments
    WiFox: Scaling WiFi Performance for Large Audience Environments ∗ ∗ Arpit Gupta, Jeongki Min, Injong Rhee Dept. of Computer Science, North Carolina State University {agupta13, jkmin, rhee}@ncsu.edu ABSTRACT 1. INTRODUCTION WiFi-based wireless LANs (WLANs) are widely used for WLANs are the most popular means of access to the In- Internet access. They were designed such that an Access ternet. The proliferation of mobile devices equipped with Points (AP) serves few associated clients with symmetric WiFi interfaces, such as smart phones, laptops, and personal uplink/downlink traffic patterns. Usage of WiFi hotspots in mobile multimedia devices, has heightened this trend. The locations such as airports and large conventions frequently performance of WiFi hotspots serving locations such as large experience poor performance in terms of downlink good- conventions and busy airports has been extremely poor. In put and responsiveness. We study the various factors re- such a setting, more than one WiFi access points (APs) sponsible for this performance degradation. We analyse provide wireless access to the Internet for many user devices and emulate a large conference network environment on our (STAs). The followings are the commonly cited causes of testbed with 45 nodes. We find that presence of asymmetry this problem. between the uplink/downlink traffic results in backlogged • packets at WiFi Access Point’s (AP’s) transmission queue Contention and collision.WhenmanySTAsare and subsequent packet losses. This traffic asymmetry re- competing for channel resources using CSMA/CA, the sults in maximum performance loss for such an environment overhead of handling high contentions, such as carrier along with degradation due to rate diversity, fairness and sensing, back-off and collisions, can be very high.
    [Show full text]
  • Network Congestion: Causes, Effects, Controls
    Data Communication & Networks G22.2262-001 Session 9 - Main Theme Network Congestion: Causes, Effects, Controls Dr. Jean-Claude Franchitti New York University Computer Science Department Courant Institute of Mathematical Sciences 1 Agenda What is Congestion? Effects of Congestion Causes/Costs of Congestion Approaches Towards Congestion Control TCP Congestion Control TCP Fairness Conclusion 2 1 Part I What is Congestion? 3 What is Congestion? Congestion occurs when the number of packets being transmitted through the network approaches the packet handling capacity of the network Congestion control aims to keep number of packets below level at which performance falls off dramatically Data network is a network of queues Generally 80% utilization is critical Finite queues mean data may be lost A top-10 problem! 4 2 Queues at a Node 5 Part II Effects of Congestion? 6 3 Effects of Congestion Packets arriving are stored at input buffers Routing decision made Packet moves to output buffer Packets queued for output transmitted as fast as possible Statistical time division multiplexing If packets arrive to fast to be routed, or to be output, buffers will fill Can discard packets Can use flow control Can propagate congestion through network 7 Interaction of Queues 8 4 Part III Causes/Costs of Congestion 9 Causes/Costs of Congestion: Scenario 1 • two senders, two receivers • one router, infinite buffers • no retransmission • large delays when congested •maximum achievable throughput 10 5 Causes/Costs of Congestion: Scenario 2
    [Show full text]
  • Instrumentation for Measuring Users' Goodputs in Dense Wi-Fi
    Instrumentation for measuring users’ goodputs in dense Wi-Fi deployments and capacity-planning rules Jos´eLuis Garc´ıa-Dorado1, Javier Ramos1, Francisco J. Gomez-Arribas1,2, Eduardo Maga˜na2,3, and Javier Aracil1,2 1 High Performance Computing and Networking research group, Universidad Aut´onoma de Madrid, Spain 2 Naudit High Performance Computing and Networking, S.L., Spain 3 Telecommunications, Networks and Services Research Group, Universidad P´ublica de Navarra, Pamplona, Spain NOTE: This is a version of an unedited manuscript that was accepted for publi- cation. Please, cite as: Jos´eLuis Garc´ıa-Dorado, Javier Ramos, Francisco J. Gomez-Arribas, Eduardo Maga˜naand Javier Aracil. Instrumentation for measuring users’ goodputs in dense Wi-Fi deployments and capacity-planning rules. Wireless Networks, Springer, vol. 26, 2943-2955, (2020). Thefinal publication is available at: https://doi.org/10.1007/s11276-019-02229-7 Abstract Before a dense Wi-Fi network is deployed, Wi-Fi providers must be careful with the perfor- mance promises they made in their way to win a bidding process. After such deployment takes place, Wi-Fi-network owners—such as public institutions—must verify that the QoS agreements are being fulfilled. We have merged both needs into a low-cost measurement system, a report of measurements at diverse scenarios and a performance prediction tool. The measurement sys- tem allows measuring the actual goodput that a set of users are receiving, and it has been used in a number of schools on a national scale. From this experience, we report measurements for different scenarios and diverse factors—which may result of interest to practitioners by them- selves.
    [Show full text]
  • A Comprehensive Overview of TCP Congestion Control in 5G Networks: Research Challenges and Future Perspectives
    sensors Review A Comprehensive Overview of TCP Congestion Control in 5G Networks: Research Challenges and Future Perspectives Josip Lorincz 1,* , Zvonimir Klarin 2 and Julije Ožegovi´c 1 1 Faculty of Electrical Engineering, Mechanical Engineering and Naval Architecture (FESB), University of Split, 21 000 Split, Croatia; [email protected] 2 Polytechnic of Sibenik, Trg Andrije Hebranga 11, 22 000 Sibenik, Croatia; [email protected] * Correspondence: [email protected] Abstract: In today’s data networks, the main protocol used to ensure reliable communications is the transmission control protocol (TCP). The TCP performance is largely determined by the used congestion control (CC) algorithm. TCP CC algorithms have evolved over the past three decades and a large number of CC algorithm variations have been developed to accommodate various network environments. The fifth-generation (5G) mobile network presents a new challenge for the implemen- tation of the TCP CC mechanism, since networks will operate in environments with huge user device density and vast traffic flows. In contrast to the pre-5G networks that operate in the sub-6 GHz bands, the implementation of TCP CC algorithms in 5G mmWave communications will be further compro- mised with high variations in channel quality and susceptibility to blockages due to high penetration losses and atmospheric absorptions. These challenges will be particularly present in environments such as sensor networks and Internet of Things (IoT) applications. To alleviate these challenges, this paper provides an overview of the most popular single-flow and multy-flow TCP CC algorithms used in pre-5G networks. The related work on the previous examinations of TCP CC algorithm Citation: Lorincz, J.; Klarin, Z.; performance in 5G networks is further presented.
    [Show full text]
  • Data Communicatoin Networking
    DATA COMMUNICATOIN NETWORKING Instructor: Ouldooz Baghban Karimi Course Book: Computer Networking, A Top-Down Approach By: Kurose, Ross Introduction Course Overview . Basics of Computer Networks . Internet & Protocol Stack . Application Layer . Transport Layer . Network Layer . Data Link Layer . Advanced Topics . Case Studies of Computer Networks . Internet Applications . Network Management . Network Security Introduction 2 Congestion . Too many sources sending too much data too fast for network to handle . Different from flow control . Manifestations . Lost packets (buffer overflow at routers) . Long delays (queuing in router buffers) Introduction 3 Causes & Costs of Congestion original data: l Scenario (1) in throughput: lout . Two senders, two receivers Host A . One router, infinite buffers unlimited shared . Output link capacity: R output link buffers . No retransmission Host B R/2 out l delay lin R/2 lin R/2 Maximum per-connection throughput: R/2 Large delays as arrival rate, lin, approaches capacity Introduction 4 Causes & Costs of Congestion Scenario (2) . One router, finite buffers . Sender retransmission of timed-out packet . Application-layer input = application-layer output: lin = lout Transport-layer input includes retransmissions : lin ≥ lin lin : original data lout l'in: original data, plus retransmitted data Host A Host B finite shared output link buffers Introduction 5 Causes & Costs of Congestion R/2 Idealization: Perfect Knowledge . Sender sends only when router buffers out available l lin R/2 lin : original data copy lout l'in: original data, plus retransmitted data Host A free buffer space! Host B finite shared output link buffers Introduction 6 Causes & Costs of Congestion Idealization: Known loss packets can be lost, dropped at router due to full buffers .
    [Show full text]
  • Nighthawk Pro Gaming Wifi 6 AX5400 Router Model XR1000
    User Manual Nighthawk Pro Gaming Router Model XR1000 NETGEAR, Inc. September 2020 350 E. Plumeria Drive 202-12095-01 San Jose, CA 95134, USA Nighthawk Pro Gaming Router Model XR1000 Support and Community Visit netgear.com/support to get your questions answered and access the latest downloads. You can also check out our NETGEAR Community for helpful advice at community.netgear.com. Regulatory and Legal Si ce produit est vendu au Canada, vous pouvez accéder à ce document en français canadien à https://www.netgear.com/support/download/. (If this product is sold in Canada, you can access this document in Canadian French at https://www.netgear.com/support/download/.) For regulatory compliance information including the EU Declaration of Conformity, visit https://www.netgear.com/about/regulatory/. See the regulatory compliance document before connecting the power supply. For NETGEAR’s Privacy Policy, visit https://www.netgear.com/about/privacy-policy. By using this device, you are agreeing to NETGEAR’s Terms and Conditions at https://www.netgear.com/about/terms-and-conditions. If you do not agree, return the device to your place of purchase within your return period. Trademarks ©NETGEAR, Inc. NETGEAR and the NETGEAR Logo are trademarks of NETGEAR, Inc. Any non-NETGEAR trademarks are used for reference purposes only. 2 Contents Chapter 1 Hardware Setup Unpack Your Router...........................................................................11 LEDs and Buttons on the Top Panel.................................................12 Back Panel...........................................................................................14
    [Show full text]
  • IBM Aspera FASP High-Speed Transport a Critical Technology Comparison to Alternative TCP-Based Transfer Technologies
    IBM Cloud White paper IBM Aspera FASP High-Speed transport A critical technology comparison to alternative TCP-based transfer technologies Introduction Contents: In this digital world, fast and reliable movement of digital data, including massive sizes over global distances, is 1 Introduction becoming vital to business success across virtually every 2 High-speed TCP overview industry. The Transmission Control Protocol (TCP) that has 4 UDP-based high-speed solutions traditionally been the engine of this data movement, however, has inherent bottlenecks in performance (Figure 1), especially for 10 IBM® Aspera® FASP® solution networks with high, round-trip time (RTT) and packet loss, and most pronounced on high-bandwidth networks. It is well understood that these inherent “soft” bottlenecks arcaused by TCP’s Additive- Increase-Multiplicative-Decrease (AIMD) congestion avoidance algorithm, which slowly probes the available bandwidth of the network, increasing the transmission rate until packet loss is detected and then exponentially reducing the transmission rate. However, it is less understood that other sources of packet loss, such as losses due to the physical network media, not associated with network congestion equally reduce the transmission rate. In fact, TCP AIMD itself creates losses, and equally contributes to the bottleneck. In ramping up the transmission rate until loss occurs, AIMD inherently overdrives the available bandwidth. In some cases, this self-induced IBM Cloud White paper loss actually surpasses loss from other causes (e.g., physical bandwidth capacities characteristic of the commodity media or bursts of cross traffic) and turns a loss-free Internet WAN environments today requires a new and communication “channel” to an unreliable “channel” with an innovative approach to bulk data movement, specifically, an unpredictable loss ratio.
    [Show full text]