57 research outputs found

    Active Queue Management for Fair Resource Allocation in Wireless Networks

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    This paper investigates the interaction between end-to-end flow control and MAC-layer scheduling on wireless links. We consider a wireless network with multiple users receiving information from a common access point; each user suffers fading, and a scheduler allocates the channel based on channel quality,but subject to fairness and latency considerations. We show that the fairness property of the scheduler is compromised by the transport layer flow control of TCP New Reno. We provide a receiver-side control algorithm, CLAMP, that remedies this situation. CLAMP works at a receiver to control a TCP sender by setting the TCP receiver's advertised window limit, and this allows the scheduler to allocate bandwidth fairly between the users

    Integration of Linux TCP and Simulation: Verification, Validation and Application

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    Network simulator has been acknowledged as one of the most flexible means in studying and developing protocol as it allows virtually endless numbers of simulated network environments to be setup and protocol of interest to be fine-tuned without requiring any real-world complicated and costly network experiment. However, depending on researchers, the same protocol of interest can be developed in different ways and different implementations may yield the outcomes that do not accurately capture the dynamics of the real protocol. In the last decade, TCP, the protocol on which the Internet is based, has been extensively studied in order to study and reevaluate its performance particularly when TCP based applications and services are deployed in an emerging Next Generation Network (NGN) and Next Generation Internet (NGI). As a result, to understand the realistic interaction of TCP with new types of networks and technologies, a combination of a real-world TCP and a network simulator seems very essential. This work presents an integration of real-world TCP implementation of Linux TCP/IP network stack into a network simulator, called INET. Moreover, verification and validation of the integrated Linux TCP are performed within INET framework to ensure the validity of the integration. The results clearly confirm that the integrated Linux TCP displays reasonable and consistent dynamics with respect to the behaviors of the real-world Linux TCP. Finally, to demonstrate the application of the INET with Linux TCP extension, algorithms of other Linux TCP variants and their dynamic over a large-bandwidth long-delay network are briefly presented

    Performance Enhancement of Multipath TCP for Wireless Communications with Multiple Radio Interfaces

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    ArticleMultipath TCP (MPTCP) allows a TCP connection to operate across multiple paths simultaneously and becomes highly attractive to support the emerging mobile devices with various radio interfaces and to improve resource utilization as well as connection robustness. The existing multipath congestion control algorithms, however, are mainly loss-based and prefer the paths with lower drop rates, leading to severe performance degradation in wireless communication systems where random packet losses occur frequently. To address this challenge, this paper proposes a new mVeno algorithm, which makes full use of the congestion information of all the subflows belonging to a TCP connection in order to adaptively adjust the transmission rate of each subflow. Specifically, mVeno modifies the additive increase phase of Veno so as to effectively couple all subflows by dynamically varying the congestion window increment based on the receiving ACKs. The weighted parameter of each subflow for tuning the congestio

    STCP: A New Transport Protocol for High-Speed Networks

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    Transmission Control Protocol (TCP) is the dominant transport protocol today and likely to be adopted in future high‐speed and optical networks. A number of literature works have been done to modify or tune the Additive Increase Multiplicative Decrease (AIMD) principle in TCP to enhance the network performance. In this work, to efficiently take advantage of the available high bandwidth from the high‐speed and optical infrastructures, we propose a Stratified TCP (STCP) employing parallel virtual transmission layers in high‐speed networks. In this technique, the AIMD principle of TCP is modified to make more aggressive and efficient probing of the available link bandwidth, which in turn increases the performance. Simulation results show that STCP offers a considerable improvement in performance when compared with other TCP variants such as the conventional TCP protocol and Layered TCP (LTCP)

    NexGen D-TCP: Next generation dynamic TCP congestion control algorithm

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    With the advancement of wireless access networks and mmWave New Radio (NR), new applications emerged, which requires a high data rate. The random packet loss due to mobility and channel conditions in a wireless network is not negligible, which degrades the significant performance of the Transmission Control Protocol (TCP). The TCP has been extensively deployed for congestion control in the communication network during the last two decades. Different variants are proposed to improve the performance of TCP in various scenarios, specifically in lossy and high bandwidth-delay product (high- BDP) networks. Implementing a new TCP congestion control algorithm whose performance is applicable over a broad range of network conditions is still a challenge. In this article, we introduce and analyze a Dynamic TCP (D-TCP) congestion control algorithm overmmWave NR and LTE-A networks. The proposed D-TCP algorithm copes up with the mmWave channel fluctuations by estimating the available channel bandwidth. The estimated bandwidth is used to derive the congestion control factor N. The congestion window is increased/decreased adaptively based on the calculated congestion control factor. We evaluated the performance of D-TCP in terms of congestion window growth, goodput, fairness and compared it with legacy and existing TCP algorithms. We performed simulations of mmWave NR during LOS \u3c-\u3e NLOS transitions and showed that D-TCP curtails the impact of under-utilization during mobility. The simulation results and live air experiment points out that D-TCP achieves 32:9% gain in goodput as compared to TCPReno and attains 118:9% gain in throughput as compared to TCP-Cubic

    A study of the effects of TCP designs on server efficiency and throughputs on wired and wireless networks.

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    Yeung, Fei-Fei.Thesis (M.Phil.)--Chinese University of Hong Kong, 2003.Includes bibliographical references (leaves 144-146).Abstracts in English and Chinese.Introduction --- p.1Chapter Part I: --- A New Socket API for Enhancing Server Efficiency --- p.5Chapter Chapter 1 --- Introduction --- p.6Chapter 1.1 --- Brief Background --- p.6Chapter 1.2 --- Deficiencies of Nagle's Algorithm and Goals and Objectives of this Research --- p.7Chapter 1.2.1 --- Effectiveness of Nagle's Algorithm --- p.7Chapter 1.2.2 --- Preventing Small Packets via Application Layer --- p.9Chapter 1.2.3 --- Minimum Delay in TCP Buffer --- p.10Chapter 1.2.4 --- Maximum Delay in TCP Buffer --- p.11Chapter 1.2.5 --- New Socket API --- p.12Chapter 1.3 --- Scope of Research and Summary of Contributions --- p.12Chapter 1.4 --- Organization of Part 1 --- p.13Chapter Chapter 2 --- Background --- p.14Chapter 2.1 --- Review of Nagle's Algorithm --- p.14Chapter 2.2 --- Additional Problems Inherent in Nagle's Algorithm --- p.17Chapter 2.3 --- Previous Proposed Modifications on Nagle's Algorithm --- p.22Chapter 2.3.1 --- The Minshall Modification --- p.22Chapter 2.3.1.1 --- The Minshall Modification --- p.22Chapter 2.3.1.2 --- The Minshall et al. Modification --- p.23Chapter 2.3.2 --- The Borman Modification --- p.23Chapter 2.3.3 --- The Jeffrey et al. Modification --- p.25Chapter 2.3.3.1 --- The EOM and MORE Variants --- p.25Chapter 2.3.3.2 --- The DLDET Variant --- p.26Chapter 2.3.4 --- Comparison Between Our Proposal and Related Works --- p.26Chapter Chapter 3 --- Min-Delay-Max-Delay TCP Buffering --- p.28Chapter 3.1 --- Minimum Delay --- p.29Chapter 3.1.1 --- Why Enabling Nagle's Algorithm Alone is Not a Solution? --- p.29Chapter 3.1.2 --- Advantages of Min-Delay TCP-layer Buffering versus Application-layer Buffering --- p.30Chapter 3.2 --- Maximum Delay --- p.32Chapter 3.2.1 --- Why Enabling Nagle's Algorithm Alone is Not a Solution? --- p.32Chapter 3.2.2 --- Advantages of Max-delay TCP Buffering versus Nagle's Algorithm --- p.33Chapter 3.3 --- Interaction with Nagle's Algorithm --- p.34Chapter 3.4 --- When to Apply Our Proposed Scheme? --- p.36Chapter 3.5 --- New Socket Option Description --- p.38Chapter 3.6 --- Implementation --- p.40Chapter 3.6.1 --- Small Packet Transmission Decision Logic --- p.42Chapter 3.6.2 --- Modified API --- p.44Chapter Chapter 4 --- Experiments --- p.46Chapter 4.1 --- The Effect of Kernel Buffering Mechanism on the Service Time --- p.47Chapter 4.1.1 --- Aims and Methodology --- p.47Chapter 4.1.2 --- Comparison of Transmission Time Required --- p.49Chapter 4.2 --- Performance of Min-Delay-Max-Delay Scheme --- p.56Chapter 4.2.1 --- Methodology --- p.56Chapter 4.2.1.1 --- Network Setup --- p.56Chapter 4.2.1.2 --- Traffic Model --- p.58Chapter 4.2.1.3 --- Delay Measurement --- p.60Chapter 4.2.2 --- Efficiency of Busy Server --- p.62Chapter 4.2.2.1 --- Performance of Nagle's algorithm --- p.62Chapter 4.2.2.2 --- Performance of Min-Delay TCP Buffering Scheme --- p.67Chapter 4.2.3 --- Limiting Delay by Setting TCP´ؤMAXDELAY --- p.70Chapter 4.3 --- Performance Sensitivity Discussion --- p.77Chapter 4.3.1 --- Sensitivity to Data Size per Invocation of send() --- p.77Chapter 4.3.2 --- Sensitivity to Minimum Delay --- p.83Chapter 4.3.3 --- Sensitivity to Round Trip Time --- p.85Chapter Chapter 5 --- Conclusion --- p.88Chapter Part II: --- Two Analytical Models for a Refined TCP Algorithm (TCP Veno) for Wired/Wireless Networks --- p.91Chapter Chapter 1 --- Introduction --- p.92Chapter 1.1 --- Brief Background --- p.92Chapter 1.2 --- Motivation and Two Analytical Models --- p.95Chapter 1.3 --- Organization of Part II --- p.96Chapter Chapter 2 --- Background --- p.97Chapter 2.1 --- TCP Veno Algorithm --- p.97Chapter 2.1.1 --- Packet Loss Type Identification --- p.97Chapter 2.1.2 --- Refined AIMD Algorithm --- p.99Chapter 2.1.2.1 --- Random Loss Management --- p.99Chapter 2.1.2.2 --- Congestion Management --- p.100Chapter 2.2 --- A Simple Model of TCP Reno --- p.101Chapter 2.3 --- Stochastic Modeling of TCP Reno over Lossy Channels --- p.103Chapter Chapter 3 --- Two Analytical Models --- p.104Chapter 3.1 --- Simple Model --- p.104Chapter 3.1.1 --- Random-loss Only Case --- p.105Chapter 3.1.2 --- Congestion-loss Only Case --- p.108Chapter 3.1.3 --- The General Case (Random + Congestion Loss) --- p.110Chapter 3.2 --- Markov Model --- p.115Chapter 3.2.1 --- Congestion Window Evolution --- p.115Chapter 3.2.2 --- Average Throughput Formulating --- p.119Chapter 3.2.2.1 --- Random-loss Only Case --- p.120Chapter 3.2.2.2 --- Congestion-loss Only Case --- p.122Chapter 3.2.2.3 --- The General Case (Random + Congestion Loss) --- p.123Chapter Chapter 4 --- Comparison with Experimental Results and Discussions --- p.127Chapter 4.1 --- Throughput versus Random Loss Probability --- p.127Chapter 4.2 --- Throughput versus Normalized Buffer Size --- p.132Chapter 4.3 --- Throughput versus Bandwidth in Asymmetric Networks --- p.135Chapter 4.3 --- Summary --- p.136Chapter Chapter 5 --- Sensitivity of TCP Veno Throughput to Various Parameters --- p.137Chapter 5.1 --- Multiplicative Decrease Factor (α) --- p.137Chapter 5.2 --- Number of Backlogs (β) and Fractional Increase Factor (γ) --- p.139Chapter Chapter 6 --- Conclusions --- p.142Bibliography --- p.14

    TCP performance enhancement in wireless networks via adaptive congestion control and active queue management

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    The transmission control protocol (TCP) exhibits poor performance when used in error-prone wireless networks. Remedy to this problem has been an active research area. However, a widely accepted and adopted solution is yet to emerge. Difficulties of an acceptable solution lie in the areas of compatibility, scalability, computational complexity and the involvement of intermediate routers and switches. This dissertation rexriews the current start-of-the-art solutions to TCP performance enhancement, and pursues an end-to-end solution framework to the problem. The most noticeable cause of the performance degradation of TCP in wireless networks is the higher packet loss rate as compared to that in traditional wired networks. Packet loss type differentiation has been the focus of many proposed TCP performance enhancement schemes. Studies conduced by this dissertation research suggest that besides the standard TCP\u27s inability of discriminating congestion packet losses from losses related to wireless link errors, the standard TCP\u27s additive increase and multiplicative decrease (AIMD) congestion control algorithm itself needs to be redesigned to achieve better performance in wireless, and particularly, high-speed wireless networks. This dissertation proposes a simple, efficient, and effective end-to-end solution framework that enhances TCP\u27s performance through techniques of adaptive congestion control and active queue management. By end-to-end, it means a solution with no requirement of routers being wireless-aware or wireless-specific . TCP-Jersey has been introduced as an implementation of the proposed solution framework, and its performance metrics have been evaluated through extensive simulations. TCP-Jersey consists of an adaptive congestion control algorithm at the source by means of the source\u27s achievable rate estimation (ARE) —an adaptive filter of packet inter-arrival times, a congestion indication algorithm at the links (i.e., AQM) by means of packet marking, and a effective loss differentiation algorithm at the source by careful examination of the congestion marks carried by the duplicate acknowledgment packets (DUPACK). Several improvements to the proposed TCP-Jersey have been investigated, including a more robust ARE algorithm, a less computationally intensive threshold marking algorithm as the AQM link algorithm, a more stable congestion indication function based on virtual capacity at the link, and performance results have been presented and analyzed via extensive simulations of various network configurations. Stability analysis of the proposed ARE-based additive increase and adaptive decrease (AJAD) congestion control algorithm has been conducted and the analytical results have been verified by simulations. Performance of TCP-Jersey has been compared to that of a perfect , but not practical, TCP scheme, and encouraging results have been observed. Finally the framework of the TCP-Jersey\u27s source algorithm has been extended and generalized for rate-based congestion control, as opposed to TCP\u27s window-based congestion control, to provide a design platform for applications, such as real-time multimedia, that do not use TCP as transport protocol yet do need to control network congestion as well as combat packet losses in wireless networks. In conclusion, the framework architecture presented in this dissertation that combines the adaptive congestion control and active queue management in solving the TCP performance degradation problem in wireless networks has been shown as a promising answer to the problem due to its simplistic design philosophy complete compatibility with the current TCP/IP and AQM practice, end-to-end architecture for scalability, and the high effectiveness and low computational overhead. The proposed implementation of the solution framework, namely TCP-Jersey is a modification of the standard TCP protocol rather than a completely new design of the transport protocol. It is an end-to-end approach to address the performance degradation problem since it does not require split mode connection establishment and maintenance using special wireless-aware software agents at the routers. The proposed solution also differs from other solutions that rely on the link layer error notifications for packet loss differentiation. The proposed solution is also unique among other proposed end-to-end solutions in that it differentiates packet losses attributed to wireless link errors from congestion induced packet losses directly from the explicit congestion indication marks in the DUPACK packets, rather than inferring the loss type based on packet delay or delay jitter as in many other proposed solutions; nor by undergoing a computationally expensive off-line training of a classification model (e.g., HMM), or a Bayesian estimation/detection process that requires estimations of a priori loss probability distributions of different loss types. The proposed solution is also scalable and fully compatible to the current practice in Internet congestion control and queue management, but with an additional function of loss type differentiation that effectively enhances TCP\u27s performance over error-prone wireless networks. Limitations of the proposed solution architecture and areas for future researches are also addressed

    A User-level, Reliable and Reconfigurable Transport Layer Protocol

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    Over the past 15 years, the Internet has proven itself to be one of the most influential inventions that humankind has ever conceived. The success of the Internet can be largely attributed to its stability and ease of access. Among the various pieces of technologies that constitute the Internet, TCP/IP can be regarded as the cornerstone to the Internet’s impressive scalability and stability. Many researchers have been and are currently actively engaged in the studies on the optimization of TCP’s performance in various network environments. This thesis presents an alternative transport layer protocol called RRTP, which is designed to provide reliable transport layer services to software applications. The motivation for this work comes from the fact that the most commonly used versions of TCP perform unsatisfactorily when they are deployed over non-conventional network platforms such as cellular/wireless, satellite, and long fat pipe networks. These non-conventional networks usually have higher network latency and link failure rate as compared with the conventional wired networks and the classic versions of TCP are unable to adapt to these characteristics. This thesis attempts to address this problem by introducing a user-level, reliable, and reconfigurable transport layer protocol that runs on top of UDP and appropriately tends to the characteristics of non-conventional networks that TCP by default ignores. A novel aspect of RRTP lies in identifying three key characteristic parameters of a network to optimize its performance. The single most important contribution of this work is its empirical demonstration of the fact that parameter-based, user-configurable, flow-control and congestion-control algorithms are highly effective at adapting to and fully utilizing various networks. This fact is demonstrated through experiments designed to benchmark the performance of RRTP against that of TCP on simulated as well as real-life networks. The experimental results indicate that the performance of RRTP consistently match and exceed TCP’s performance on all major network platforms. This leads to the conclusion that a user-level, reliable, and reconfigurable transport-layer protocol, which possesses the essential characteristics of RRTP, would serve as a viable replacement for TCP over today’s heterogeneous network platforms
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