Improved transmission control protocol congestion control technique for high bandwidth long distance networks

Transmission Control Protocol (TCP) is responsible for reliable communication of data in high bandwidth long distance networks. TCP is reliable because of its congestion control technique. Many TCP congestion control techniques for different operating systems have been developed previously. TCP Compound and TCP CUBIC are current congestion control techniques being used in Microsoft Windows and Linux operating systems respectively. TCP Reno is Standard TCP congestion control technique. TCP CUBIC does not perform well in high bandwidth long distance networks due to its exponential growth and less reduction in congestion window size. This leads to burst packet losses, unfair allocation of unused link bandwidth, long convergence time, and poor TCP friendliness among competing flows. The aim of this research work is to develop an improved congestion control technique based on TCP CUBIC for high bandwidth long distance networks. This improved technique is based on three components which are Congestion Control Technique for Slow Start (CCT-SS), Congestion Control Technique for Loss Occurrence (CCT-LO), and Enhanced Response Function of TCP CUBIC (ERFC). CCT-SS is proposed which increases the lower boundary limit of congestion window, which in turn, decreases the packet loss rate. CCT-LO is proposed which introduces a new congestion window reduction parameter in order to achieve fairer and quicker allocation of link bandwidth among the competing flows. ERFC is proposed which reduces the average congestion window size of TCP CUBIC in order to improve the TCP friendliness. As a conjunctive result of this research work, an improved congestion control technique is developed by combining the CCT-SS, CCT-LO and ERFC components. Network Simulator 2 is used to evaluate the performance of the proposed congestion control technique and to compare it with the current and other congestion control techniques. Results show that the performance of the proposed congestion control technique outperforms by 8.4% as compared to current congestion control technique

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

Model based analysis of some high speed network issues

The study of complex problems in science and engineering today typically involves large scale data, huge number of large-scale scientific breakthroughs critically depends on large multi-disciplinary and geographically-dispersed research teams, where the high speed network becomes the integral part. To serve the ongoing bandwidth requirement and scalability of these networks, there has been a continuous evolution of different TCPs for high speed networks. Testing these protocols on a real network would be expensive, time consuming and more over not easily available to the researchers worldwide. Network simulation is well accepted and widely used method for performance evaluation, it is well known that packet-based simulators like NS2 and Opnet are not adequate in high speed also in large scale networks because of its inherent bottlenecks in terms of message overhead and execution time. In that case model based approach with the help of a set of coupled differential equations is preferred for simulations. This dissertation is focused on the key challenges on research and development of TCPs on high-speed network. To address these issues/challenges this thesis has three objectives: design an analytical simulation methodology; model behaviors of high speed networks and other components including TCP flows and queue using the analytical simulation method; analyze them and explore impacts and interrelationship among them. To decrease the simulation time and speed up the process of testing and development of high speed TCP, we present a scalable simulation methodology for high speed network. We present the fluid model equations for various high-speed TCP variants. With the help of these fluid model equations, the behavior of high-speed TCP variants under various scenarios and its effect on queue size variations are presented. High speed network is not feasible unless we understand effect of bottleneck buffer size on performance of these high-speed TCP variants. A fluid model is introduced to accommodate the new observations of synchronization and de-synchronization phenomena of packet losses at bottleneck link and a microscopic analysis is presented on different buffer sizes at drop-tail queuing scheme. The proposed model based methods promotes principal understanding of the future heterogeneous networks and accelerates protocol developments

A User-level, Reliable and Reconfigurable Transport Layer Protocol

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

Congestion Control for Streaming Media

The Internet has assumed the role of the underlying communication network for applications such as file transfer, electronic mail, Web browsing and multimedia streaming. Multimedia streaming, in particular, is growing with the growth in power and connectivity of today\u27s computers. These Internet applications have a variety of network service requirements and traffic characteristics, which presents new challenges to the single best-effort service of today\u27s Internet. TCP, the de facto Internet transport protocol, has been successful in satisfying the needs of traditional Internet applications, but fails to satisfy the increasingly popular delay sensitive multimedia applications. Streaming applications often use UDP without a proper congestion avoidance mechanisms, threatening the well-being of the Internet. This dissertation presents an IP router traffic management mechanism, referred to as Crimson, that can be seamlessly deployed in the current Internet to protect well-behaving traffic from misbehaving traffic and support Quality of Service (QoS) requirements of delay sensitive multimedia applications as well as traditional Internet applications. In addition, as a means to enhance Internet support for multimedia streaming, this dissertation report presents design and evaluation of a TCP-Friendly and streaming-friendly transport protocol called the Multimedia Transport Protocol (MTP). Through a simulation study this report shows the Crimson network efficiently handles network congestion and minimizes queuing delay while providing affordable fairness protection from misbehaving flows over a wide range of traffic conditions. In addition, our results show that MTP offers streaming performance comparable to that provided by UDP, while doing so under a TCP-Friendly rate

Improved algorithms for TCP congestion control

Reliable and efficient data transfer on the Internet is an important issue. Since late 70’s the protocol responsible for that has been the de facto standard TCP, which has proven to be successful through out the years, its self-managed congestion control algorithms have retained the stability of the Internet for decades. However, the variety of existing new technologies such as high-speed networks (e.g. fibre optics) with high-speed long-delay set-up (e.g. cross-Atlantic links) and wireless technologies have posed lots of challenges to TCP congestion control algorithms. The congestion control research community proposed solutions to most of these challenges. This dissertation adds to the existing work by: firstly tackling the highspeed long-delay problem of TCP, we propose enhancements to one of the existing TCP variants (part of Linux kernel stack). We then propose our own variant: TCP-Gentle. Secondly, tackling the challenge of differentiating the wireless loss from congestive loss in a passive way and we propose a novel loss differentiation algorithm which quantifies the noise in packet inter arrival times and use this information together with the span (ratio of maximum to minimum packet inter arrival times) to adapt the multiplicative decrease factor according to a predefined logical formula. Finally, extending the well-known drift model of TCP to account for wireless loss and some hypothetical cases (e.g. variable multiplicative decrease), we have undertaken stability analysis for the new version of the model