STCP: A New Transport Protocol for High-Speed Networks

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)

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

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

Active Queue Management for Fair Resource Allocation in Wireless Networks

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

TCP FTAT (Fast Transmit Adaptive Transmission): a New End-To-End Congestion Control Algorithm

Congestion Control in TCP is the algorithm that controls allocation of network resources for a number of competing users sharing a network. The nature of computer networks, which can be described from the TCP protocol perspective as unknown resources for unknown traffic of users, means that the functionality of the congestion control algorithm in TCP requires explicit feedback from the network on which it operates. Unfortunately this is not the way it works with TCP, as one of the fundamental principles of the TCP protocol is to be end-to-end, in order to be able to operate on any network, which can consist of hundreds of routers and hundreds of links with varying bandwidth and capacities. This fact requires the Congestion Control algorithm to be adaptive by nature, to adapt to the network environment under any given circumstances and to obtain the required feedback implicitly through observation and measurements. In this thesis we propose a new TCP end-to-end congestion control algorithm that provides performance improvements over existing TCP congestion control algorithms in computer networks in general, and an even greater improvement in wireless and/or high bandwidth- delay product network

TCP FTAT (Fast Transmit Adaptive Transmission): a New End-To-End Congestion Control Algorithm

Congestion Control in TCP is the algorithm that controls allocation of network resources for a number of competing users sharing a network. The nature of computer networks, which can be described from the TCP protocol perspective as unknown resources for unknown traffic of users, means that the functionality of the congestion control algorithm in TCP requires explicit feedback from the network on which it operates. Unfortunately this is not the way it works with TCP, as one of the fundamental principles of the TCP protocol is to be end-to-end, in order to be able to operate on any network, which can consist of hundreds of routers and hundreds of links with varying bandwidth and capacities. This fact requires the Congestion Control algorithm to be adaptive by nature, to adapt to the network environment under any given circumstances and to obtain the required feedback implicitly through observation and measurements. In this thesis we propose a new TCP end-to-end congestion control algorithm that provides performance improvements over existing TCP congestion control algorithms in computer networks in general, and an even greater improvement in wireless and/or high bandwidth- delay product network

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

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

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

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

SACK TCP VENO

by Chung Ling Chi.Thesis (M.Phil.)--Chinese University of Hong Kong, 2001.Includes bibliographical references (leaves 74-76).Abstracts in English and Chinese.Chapter Chapter 1 --- Introduction --- p.1Chapter 1.1 --- Overview --- p.1Chapter 1.2 --- Motivation and Proposed Solution --- p.2Chapter 1.3 --- Organization of the Thesis --- p.4Chapter Chapter 2 --- Background --- p.5Chapter 2.1 --- Basics of Transmission Control Protocol --- p.5Chapter 2.1.1 --- Slow Start and Congestion Avoidance --- p.5Chapter 2.1.2 --- Fast Retransmit and Fast Recovery --- p.7Chapter 2.2 --- SACK TCP Mechanism --- p.8Chapter 2.2.1 --- SACK-permitted Option during Three-way Handshake --- p.8Chapter 2.2.2 --- SACK blocks in SACK Option --- p.9Chapter 2.2.3 --- Interpreting the SACK Option using Scoreboard --- p.10Chapter 2.2.4 --- Retransmission Strategy --- p.11Chapter 2.3 --- TCP Veno Mechanism --- p.13Chapter 2.3.1 --- Refined Additive Increase --- p.13Chapter 2.3.2 --- Refined Multiplicative Decrease --- p.14Chapter Chapter 3 --- SACK TCPVeno --- p.16Chapter 3.1 --- Distinguishing between Types of Packet Loss --- p.17Chapter 3.2 --- Refined Multiplicative Decrease --- p.21Chapter 3.2.1 --- Algorithm --- p.21Chapter 3.2.2 --- Recovery in Consecutive packet Losses --- p.22Chapter 3.2.3 --- Recovering Multiple Packet Losses within a Single Window --- p.26Chapter 3.3 --- Refined Additive Increase --- p.37Chapter 3.3.1 --- Algorithm --- p.37Chapter 3.3.2 --- Advantages --- p.40Chapter 3.4 --- Other Issues --- p.43Chapter 3.4.1 --- Two Side Modifications --- p.43Chapter Chapter 4 --- Experiments --- p.44Chapter 4.1 --- The Network Scenario --- p.44Chapter 4.1.1 --- Dummynet --- p.45Chapter 4.2 --- Experiment Results --- p.47Chapter 4.2.1 --- Single Connection --- p.47Chapter 4.2.1.1 --- Congestion Window Evolution --- p.47Chapter 4.2.1.2 --- Sending Rate and Throughput Evolution --- p.49Chapter 4.2.1.2.1 --- Impact of Packet Loss Rate Due to Lossy Link --- p.49Chapter 4.2.1.2.2 --- Impact of Buffering --- p.52Chapter 4.2.1.2.3 --- Impact of Propagation Delay --- p.57Chapter 4.2.2 --- Multiple Connections --- p.62Chapter 4.2.2.1 --- Fairness --- p.62Chapter 4.2.2.2 --- Compatibility --- p.67Chapter Chapter 5 --- Conclusion --- p.72Bibliography --- p.7