Impact of TCP Congestion Control on Bufferbloat in Cellular Networks
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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. -
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]. -
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. -
TCP-Friendly Congestion Control for Real-Time Streaming Applications
MIT Technical Report MIT-LCS-TR-806, May 2000. TCP-friendly Congestion Control for Real-time Streaming Applications Deepak Bansal and Hari Balakrishnan M.I.T. Laboratory for Computer Science Cambridge, MA 02139 g Email: fbansal,hari @lcs.mit.edu Abstract ment in the Internet, however, the protocols used by these applications must implement congestion control algorithms This paper introduces and analyzes a class of nonlinear con- that are stable and interact well with TCP. Such protocols gestion control algorithms called binomial algorithms, moti- are called “TCP compatible” [3] or “TCP fair”. They ensure vated in part by the needs of streaming audio and video ap- that the TCP connections using AIMD get their fair allo- plications for which a drastic reduction in transmission rate cation of bandwidth in the presence of these protocols and upon congestion is problematic. Binomial algorithms gen- vice versa. One notion that has been proposed to capture eralize TCP-style additive-increase by increasing inversely “TCP compatibility” is “TCP-friendliness”. It is well known k proportional to a power of the current window (for TCP, that the throughput of a flow with TCP’s AIMD conges- k =¼ = ½ ) ; they generalize TCP-style multiplicative-decrease tion control (increase factor « packet, decrease factor Ô Ð =½=¾ Ô » Ë=´Ê Ôµ by decreasing proportional to a power of the current win- ¬ ) is related to its loss rate as ,where Ð = ½ dow (for TCP, ). We show that there are an infinite Ë is the packet size [19, 25, 10, 26]. An algorithm is TCP- Ô » Ë=´Ê Ôµ number of deployable TCP-friendly binomial algorithms, all friendly [20] if its throughput with the same · Ð =½ of which satisfy k , and that all binomial algorithms constant of proportionality as for a TCP connection with the converge to fairness under a synchronized-feedback assump- same packet size and round-trip time. -
Better Throughput in TCP Congestion Control Algorithms on Manets
(IJACSA) International Journal of Advanced Computer Science and Applications, Special Issue on Wireless & Mobile Networks Scalable TCP: Better Throughput in TCP Congestion Control Algorithms on MANETs M.Jehan Dr. G.Radhamani Associate Professor, Department of Computer Science, Professor & Director, Department of Computer Science, D.J.Academy for Managerial Excellence, Dr.G.R.Damodaran College of Science, Coimbatore, India Coimbatore, India Abstract—In the modern mobile communication world the This paper is entirely devoted to evaluating the Control congestion control algorithms role is vital to data transmission Window (cwnd), Round Trip Delay Time (rtt) and Throughput between mobile devices. It provides better and reliable using the TCP BIC, Vegas and Scalable TCP congestion communication capabilities in all kinds of networking control algorithms in the wireless networks. environment. The wireless networking technology and the new kind of requirements in communication systems needs some II. BACKGROUND WORK extensions to the original design of TCP for on coming technology development. This work aims to analyze some TCP congestion A. Congestion Control in Transmission Control Protocol control algorithms and their performance on Mobile Ad-hoc Algorithms Networks (MANET). More specifically, we describe performance TCP (Transmission Control Protocol) is a set of rules behavior of BIC, Vegas and Scalable TCP congestion control (protocol) used along with the Internet Protocol (IP) to send algorithms. The evaluation is simulated through Network data in the form of message units between computers over the Simulator (NS2) and the performance of these algorithms is Internet. It operates at a higher level, concerned only with the analyzed in the term of efficient data transmission in wireless and two end systems. -
Bufferbloat: Advertently Defeated a Critical TCP Con- Gestion-Detection Mechanism, with the Result Being Worsened Congestion and Increased Latency
practice Doi:10.1145/2076450.2076464 protocol the presence of congestion Article development led by queue.acm.org and thus the need for compensating adjustments. Because memory now is significant- A discussion with Vint Cerf, Van Jacobson, ly cheaper than it used to be, buffering Nick Weaver, and Jim Gettys. has been overdone in all manner of net- work devices, without consideration for the consequences. Manufacturers have reflexively acted to prevent any and all packet loss and, by doing so, have in- BufferBloat: advertently defeated a critical TCP con- gestion-detection mechanism, with the result being worsened congestion and increased latency. Now that the problem has been di- What’s Wrong agnosed, people are working feverishly to fix it. This case study considers the extent of the bufferbloat problem and its potential implications. Working to with the steer the discussion is Vint Cerf, popu- larly known as one of the “fathers of the Internet.” As the co-designer of the TCP/IP protocols, Cerf did indeed play internet? a key role in developing the Internet and related packet data and security technologies while at Stanford Univer- sity from 1972−1976 and with the U.S. Department of Defense’s Advanced Research Projects Agency (DARPA) from 1976−1982. He currently serves as Google’s chief Internet evangelist. internet DeLays nOw are as common as they are Van Jacobson, presently a research maddening. But that means they end up affecting fellow at PARC where he leads the networking research program, is also system engineers just like all the rest of us. And when central to this discussion. -
Copa: Practical Delay-Based Congestion Control for the Internet
Copa: Practical Delay-Based Congestion Control for the Internet Venkat Arun and Hari Balakrishnan, MIT CSAIL https://www.usenix.org/conference/nsdi18/presentation/arun This paper is included in the Proceedings of the 15th USENIX Symposium on Networked Systems Design and Implementation (NSDI ’18). April 9–11, 2018 • Renton, WA, USA ISBN 978-1-939133-01-4 Open access to the Proceedings of the 15th USENIX Symposium on Networked Systems Design and Implementation is sponsored by USENIX. Copa: Practical Delay-Based Congestion Control for the Internet Venkat Arun and Hari Balakrishnan M.I.T. Computer Science and Artificial Intelligence Laboratory Email: fvenkatar,[email protected] Abstract time or low interactive delay). Larger BDPs exacerbate This paper introduces Copa, an end-to-end conges- the \bufferbloat" problem. A more global Internet tion control algorithm that uses three ideas. First, it leads to flows with very different propagation delays sharing a bottleneck (exacerbating the RTT-unfairness shows that a target rate equal to 1=(ddq), where dq is the (measured) queueing delay, optimizes a natural func- exhibited by many current protocols). tion of throughput and delay under a Markovian packet At the same time, application providers and users arrival model. Second, it adjusts its congestion window have become far more sensitive to performance, with in the direction of this target rate, converging quickly to notions of \quality of experience" for real-time and the correct fair rates even in the face of significant flow streaming media, and various metrics to measure Web churn. These two ideas enable a group of Copa flows performance being developed. -
Performance Analysis of Google Congestion Control Algorithm for Webrtc
Performance Analysis of Google Congestion Control Algorithm for WebRTC M.L. Guerrero Viveros Performance Analysis of Google Congestion Control Algorithm for WebRTC by M.L. Guerrero Viveros in partial fulfilment of the requirements for the degree of Master of Science in Electrical Engineering Track Telecommunications and sensing systems at the Delft University of Technology, to be defended publicly on Thursday November 07, 2019 at 10:00 AM. Student number: 4736605 Project duration: January 1, 2019 – November 07, 2019 Thesis committee: Prof. dr. ir. P. Van Mieghem, TU Delft Dr. P. Pawelczak, TU Delft Ir. R. Noldus, Ericsson, supervisor This thesis is confidential and cannot be made public until November 07, 2019. An electronic version of this thesis is available at http://repository.tudelft.nl/. Abstract Web Real-Time communication (WebRTC) is a technology that enables web browsers to es- tablish real-time communication services without the need of specific software or plug-ins. This technology is gaining popularity and is already supported by popular browsers such as Google Chrome, Firefox and Safari. The quality of real-time communication services depends highly on latency. For this reason, real-time flows have different requirements than conven- tional TCP flows which focus mainly on the transfer of bulk traffic. The IETF created the working group RMCAT (RTP Media Congestion Avoidance Techniques) to define requirements for real-time congestion control algorithms. One of the proposed algorithms is Google Con- gestion Control (GCC). This is the only real-time congestion control algorithm implemented in commercial browsers such as Google Chrome. Unfortunately, the performance of GCC in wireless networks has not been extensively evaluated.