Mitigating Bufferbloat with Receiver-Based TCP Flow Control Mechanism in Cellular Networks
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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. -
Lecture 8: Overview of Computer Networking Roadmap
Lecture 8: Overview of Computer Networking Slides adapted from those of Computer Networking: A Top Down Approach, 5th edition. Jim Kurose, Keith Ross, Addison-Wesley, April 2009. Roadmap ! what’s the Internet? ! network edge: hosts, access net ! network core: packet/circuit switching, Internet structure ! performance: loss, delay, throughput ! media distribution: UDP, TCP/IP 1 What’s the Internet: “nuts and bolts” view PC ! millions of connected Mobile network computing devices: server Global ISP hosts = end systems wireless laptop " running network apps cellular handheld Home network ! communication links Regional ISP " fiber, copper, radio, satellite access " points transmission rate = bandwidth Institutional network wired links ! routers: forward packets (chunks of router data) What’s the Internet: “nuts and bolts” view ! protocols control sending, receiving Mobile network of msgs Global ISP " e.g., TCP, IP, HTTP, Skype, Ethernet ! Internet: “network of networks” Home network " loosely hierarchical Regional ISP " public Internet versus private intranet Institutional network ! Internet standards " RFC: Request for comments " IETF: Internet Engineering Task Force 2 A closer look at network structure: ! network edge: applications and hosts ! access networks, physical media: wired, wireless communication links ! network core: " interconnected routers " network of networks The network edge: ! end systems (hosts): " run application programs " e.g. Web, email " at “edge of network” peer-peer ! client/server model " client host requests, receives -
TCP Congestion Control: Overview and Survey of Ongoing Research
Purdue University Purdue e-Pubs Department of Computer Science Technical Reports Department of Computer Science 2001 TCP Congestion Control: Overview and Survey Of Ongoing Research Sonia Fahmy Purdue University, [email protected] Tapan Prem Karwa Report Number: 01-016 Fahmy, Sonia and Karwa, Tapan Prem, "TCP Congestion Control: Overview and Survey Of Ongoing Research" (2001). Department of Computer Science Technical Reports. Paper 1513. https://docs.lib.purdue.edu/cstech/1513 This document has been made available through Purdue e-Pubs, a service of the Purdue University Libraries. Please contact [email protected] for additional information. TCP CONGESTION CONTROL: OVERVIEW AND SURVEY OF ONGOING RESEARCH Sonia Fahmy Tapan Prem Karwa Department of Computer Sciences Purdue University West Lafayette, IN 47907 CSD TR #01-016 September 2001 TCP Congestion Control: Overview and Survey ofOngoing Research Sonia Fahmy and Tapan Prem Karwa Department ofComputer Sciences 1398 Computer Science Building Purdue University West Lafayette, IN 47907-1398 E-mail: {fahmy,tpk}@cs.purdue.edu Abstract This paper studies the dynamics and performance of the various TCP variants including TCP Tahoe, Reno, NewReno, SACK, FACK, and Vegas. The paper also summarizes recent work at the lETF on TCP im plementation, and TCP adaptations to different link characteristics, such as TCP over satellites and over wireless links. 1 Introduction The Transmission Control Protocol (TCP) is a reliable connection-oriented stream protocol in the Internet Protocol suite. A TCP connection is like a virtual circuit between two computers, conceptually very much like a telephone connection. To maintain this virtual circuit, TCP at each end needs to store information on the current status of the connection, e.g., the last byte sent. -
Bit & Baud Rate
What’s The Difference Between Bit Rate And Baud Rate? Apr. 27, 2012 Lou Frenzel | Electronic Design Serial-data speed is usually stated in terms of bit rate. However, another oft- quoted measure of speed is baud rate. Though the two aren’t the same, similarities exist under some circumstances. This tutorial will make the difference clear. Table Of Contents Background Bit Rate Overhead Baud Rate Multilevel Modulation Why Multiple Bits Per Baud? Baud Rate Examples References Background Most data communications over networks occurs via serial-data transmission. Data bits transmit one at a time over some communications channel, such as a cable or a wireless path. Figure 1 typifies the digital-bit pattern from a computer or some other digital circuit. This data signal is often called the baseband signal. The data switches between two voltage levels, such as +3 V for a binary 1 and +0.2 V for a binary 0. Other binary levels are also used. In the non-return-to-zero (NRZ) format (Fig. 1, again), the signal never goes to zero as like that of return- to-zero (RZ) formatted signals. 1. Non-return to zero (NRZ) is the most common binary data format. Data rate is indicated in bits per second (bits/s). Bit Rate The speed of the data is expressed in bits per second (bits/s or bps). The data rate R is a function of the duration of the bit or bit time (TB) (Fig. 1, again): R = 1/TB Rate is also called channel capacity C. If the bit time is 10 ns, the data rate equals: R = 1/10 x 10–9 = 100 million bits/s This is usually expressed as 100 Mbits/s. -
A QUIC Implementation for Ns-3
This paper has been submitted to WNS3 2019. Copyright may be transferred without notice. A QUIC Implementation for ns-3 Alvise De Biasio, Federico Chiariotti, Michele Polese, Andrea Zanella, Michele Zorzi Department of Information Engineering, University of Padova, Padova, Italy e-mail: {debiasio, chiariot, polesemi, zanella, zorzi}@dei.unipd.it ABSTRACT One of the most important novelties is QUIC, a transport pro- Quick UDP Internet Connections (QUIC) is a recently proposed tocol implemented at the application layer, originally proposed by transport protocol, currently being standardized by the Internet Google [8] and currently considered for standardization by the Engineering Task Force (IETF). It aims at overcoming some of the Internet Engineering Task Force (IETF) [6]. QUIC addresses some shortcomings of TCP, while maintaining the logic related to flow of the issues that currently affect transport protocols, and TCP in and congestion control, retransmissions and acknowledgments. It particular. First of all, it is designed to be deployed on top of UDP supports multiplexing of multiple application layer streams in the to avoid any issue with middleboxes in the network that do not same connection, a more refined selective acknowledgment scheme, forward packets from protocols other than TCP and/or UDP [12]. and low-latency connection establishment. It also integrates cryp- Moreover, unlike TCP, QUIC is not integrated in the kernel of the tographic functionalities in the protocol design. Moreover, QUIC is Operating Systems (OSs), but resides in user space, so that changes deployed at the application layer, and encapsulates its packets in in the protocol implementation will not require OS updates. Finally, UDP datagrams. -
Buffer De-Bloating in Wireless Access Networks
Buffer De-bloating in Wireless Access Networks by Yuhang Dai A thesis submitted to the University of London for the degree of Doctor of Philosophy School of Electronic Engineering & Computer Science Queen Mary University of London United Kingdom Sep 2018 TO MY FAMILY Abstract Excessive buffering brings a new challenge into the networks which is known as Bufferbloat, which is harmful to delay sensitive applications. Wireless access networks consist of Wi-Fi and cellular networks. In the thesis, the performance of CoDel and RED are investigated in Wi-Fi networks with different types of traffic. Results show that CoDel and RED work well in Wi-Fi networks, due to the similarity of protocol structures of Wi-Fi and wired networks. It is difficult for RED to tune parameters in cellular networks because of the time-varying channel. CoDel needs modifications as it drops the first packet of queue and thehead packet in cellular networks will be segmented. The major contribution of this thesis is that three new AQM algorithms tailored to cellular networks are proposed to alleviate large queuing delays. A channel quality aware AQM is proposed using the CQI. The proposed algorithm is tested with a single cell topology and simulation results show that the proposed algo- rithm reduces the average queuing delay for each user by 40% on average with TCP traffic compared to CoDel. A QoE aware AQM is proposed for VoIP traffic. Drops and delay are monitored and turned into QoE by mathematical models. The proposed algorithm is tested in NS3 and compared with CoDel, and it enhances the QoE of VoIP traffic and the average end- to-end delay is reduced by more than 200 ms when multiple users with different CQI compete for the wireless channel. -
An Experimental Evaluation of Low Latency Congestion Control for Mmwave Links
An Experimental Evaluation of Low Latency Congestion Control for mmWave Links Ashutosh Srivastava, Fraida Fund, Shivendra S. Panwar Department of Electrical and Computer Engineering, NYU Tandon School of Engineering Emails: fashusri, ffund, [email protected] Abstract—Applications that require extremely low latency are −40 expected to be a major driver of 5G and WLAN networks that −45 include millimeter wave (mmWave) links. However, mmWave −50 links can experience frequent, sudden changes in link capacity −55 due to obstructions in the signal path. These dramatic variations RSSI (dBm) in link capacity cause a temporary “bufferbloat” condition −60 0 25 50 75 100 during which delay may increase by a factor of 2-10. Low Time (s) latency congestion control protocols, which manage bufferbloat Fig. 1: Effect of human blocker on received signal strength of by minimizing queue occupancy, represent a potential solution a 60 GHz WLAN link. to this problem, however their behavior over links with dramatic variations in capacity is not well understood. In this paper, we explore the behavior of two major low latency congestion control protocols, TCP BBR and TCP Prague (as part of L4S), using link Link rate: 5 packets/ms traces collected over mmWave links under various conditions. 1ms queuing delay Our evaluation reveals potential problems associated with use of these congestion control protocols for low latency applications over mmWave links. Link rate: 1 packet/ms Index Terms—congestion control, low latency, millimeter wave 5ms queuing delay Fig. 2: Illustration of effect of variations in link capacity on I. INTRODUCTION queueing delay. Low latency applications are envisioned to play a significant role in the long-term growth and economic impact of 5G This effect is of great concern to low-delay applications, and future WLAN networks [1]. -
LTE-Advanced
Table of Contents INTRODUCTION........................................................................................................ 5 EXPLODING DEMAND ............................................................................................... 8 Smartphones and Tablets ......................................................................................... 8 Application Innovation .............................................................................................. 9 Internet of Things .................................................................................................. 10 Video Streaming .................................................................................................... 10 Cloud Computing ................................................................................................... 11 5G Data Drivers ..................................................................................................... 11 Global Mobile Adoption ........................................................................................... 11 THE PATH TO 5G ..................................................................................................... 15 Expanding Use Cases ............................................................................................. 15 1G to 5G Evolution ................................................................................................. 17 5G Concepts and Architectures ................................................................................ 20 Information-Centric -
RTM-100 Troposcatter Modem Improved Range, Stability and Throughput for Troposcatter Communications
RTM-100 Troposcatter Modem Improved range, stability and throughput for troposcatter communications The Raytheon RTM-100 troposcatter modem sets new milestones in troposcatter communications featuring 100 MB throughput. Benefits Superior Performance The use of turbo-coding An industry first, the waveform forward error correction The RTM-100 comes as n Operation up to 100 Mbps of Raytheon’s RTM-100 (FEC) and state-of-the-art a compact 2U rack with troposcatter modem offers digital processing ensures an n Unique waveform for multipath standard 70 MHz inputs/ strong resiliency to multipath, unprecedented throughput cancellation and optimized outputs. algorithms for fading immunity which negatively affects up to 100 Mbps. The modem n Quad diversity with soft decision troposcatter communications. integrates a non-linear digital algorithm Combined with an optimized pre-distortion capability. time-interleaving process and n Highly spectrum-efficient FEC The Gigabit Ethernet (GbE) signal channel diversity, the processing data port and the ability RTM- 100 delivers superior n Dual transmission path with to command and control transmission performance. independent digital pre- the operation over Simple The sophisticated algorithms, distortion Network Management including Doppler n Ethernet data and control Protocol (SNMP) simplify the compensation and maximum interface (SNMP and Web integration of the device into ratio combining between the graphical user interface) a net-centric Internet protocol four diversity inputs, ensure n Compact 2U 19-inch -
AQM Algorithms and Their Interaction with TCP Congestion Control Mechanisms
View metadata, citation and similar papers at core.ac.uk brought to you by CORE provided by Universidad Carlos III de Madrid e-Archivo Grado Universitario en Ingenier´ıaTelem´atica 2016/2017 Trabajo Fin de Grado Control de Congesti´onTCP y mecanismos AQM Sergio Maeso Jim´enez Tutor/es Celeste Campo V´azquez Carlos Garc´ıaRubio Legan´es,2 de Octubre de 2017 Esta obra se encuentra sujeta a la licencia Creative Commons Reconocimiento - No Comercial - Sin Obra Derivada Control de Congesti´onTCP y mecanismos AQM By Sergio Maeso Jim´enez Directed By Celeste Campo V´azquez Carlos Garc´ıaRubio A Dissertation Submitted to the Department of Telematic Engineering in Partial Fulfilment of the Requirements for the BACHELOR'S DEGREE IN TELEMATICS ENGINEERING Approved by the Supervising Committee: Chairman Marta Portela Garc´ıa Chair Carlos Alario Hoyos Secretary I~naki Ucar´ Marqu´es Deputy Javier Manuel Mu~noz Garc´ıa Grade: Legan´es,2 de Octubre de 2017 iii iv Acknowledgements I would like to thanks my tutors Celeste Campo and Carlos Garcia for all the support they gave me while I was doing this thesis with them. To my parents, who believe in me against all odds. v vi Abstract In recent years, the relevance of delay over throughput has been particularly emphasized. Nowadays our networks are getting more and more sensible to latency due to the proliferation of applications and services like VoIP, IPTV or online gaming where a low delay is essential for a proper performance and a good user experience. Most of this unnecessary delay is created by the misbehaviour of many buffers that populate Internet. -
TCP Congestion Control with a Misbehaving Receiver
TCP Congestion Control with a Misbehaving Receiver Stefan Savage, Neal Cardwell, David Wetherall, and Tom Anderson Department of Computer Science and Engineering University of Washington, Seattle Abstract col modifications, a faulty or malicious receiver can at most cause the sender to transmit data at a slower rate than it otherwise would, In this paper, we explore the operation of TCP congestion control thus harming only itself. Because our work has serious practical when the receiver can misbehave, as might occur with a greedy ramifications for an Internet that depends on trust to avoid conges- Web client. We first demonstrate that there are simple attacks that tion collapse, we also describe backwards-compatible mechanisms allow a misbehaving receiver to drive a standard TCP sender ar- that can be implemented at the sender to mitigate the effects of un- bitrarily fast, without losing end-to-end reliability. These attacks trusted receivers. are widely applicable because they stem from the sender behavior As far as we are aware, the division of trust between sender specified in RFC 2581 rather than implementation bugs. We then and receiver has not been studied previously in the context of con- show that it is possible to modify TCP to eliminate this undesir- gestion control. While end-to-end congestion control protocols as- able behavior entirely, without requiring assumptions of any kind sume that both sender and receiver behave correctly, in many en- about receiver behavior. This is a strong result: with our solution vironments the interests of sender and receiver may differ consid- a receiver can only reduce the data transfer rate by misbehaving, erably – creating significant incentives to violate this “good faith” thereby eliminating the incentive to do so. -
Performance Analysis of Receive-Side Real-Time Congestion Control for Webrtc
Performance Analysis of Receive-Side Real-Time Congestion Control for WebRTC Varun Singh Albert Abello Lozano J¨org Ott Aalto University, Finland Aalto University, Finland Aalto University, Finland varun.singh@aalto.fi albert.abello.lozano@aalto.fi jorg.ott@aalto.fi Abstract—In the forthcoming deployments of WebRTC sys- congestion control, the IETF has chartered a new working tems, we speculate that high quality video conferencing will see group, RMCAT3, to standardize congestion-control for real- wide adoption. It is currently being deployed on Google Chrome time communication, which is expected to be a multi-year and Firefox web-browsers, meanwhile desktop and mobile clients are under development. Without a standardized signaling mech- process [1]; but early implementations are already available. anism, service providers can enable various types of topologies; In this paper, we evaluate the performance of WebRTC ranging from full-mesh to centralized video conferencing and video calls over different topologies and with varying amounts everything in between. In this paper, we evaluate the performance of cross-traffic. All experiments are conducted using the of various topologies using endpoints implementing WebRTC. Chrome browser and our testbed. With the testbed we are able We specifically evaluate the performance of the congestion con- trol currently implemented and deployed in these web-browser, to control the bottleneck link capacity, the end-to-end latency, Receive-side Real-Time Congestion Control (RRTCC). We use link loss rate and the queue size of intermediate routers. transport impairments like varying throughput, loss and delay, Consequently, in this testbed we can investigate the following: and varying amounts of cross-traffic to measure the performance.