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

TCP Sintok: Transmission control protocol with delay-based loss detection and contention avoidance mechanisms for mobile ad hoc networks

Mobile Ad hoc Network (MANET) consists of mobile devices that are connected to each other using a wireless channel, forming a temporary network without the aid of fixed infrastructure; in which hosts are free to move randomly as well as free to join or leave. This decentralized nature of MANET comes with new challenges that violate the design concepts of Transmission Control Protocol (TCP); the current dominant protocol of the Internet. TCP always infers packet loss as an indicator of network congestion and causes it to perform a sharp reduction to its sending rate. MANET suffers from several types of packet losses due to its mobility feature and contention on wireless channel access and these would lead to poor TCP performance. This experimental study investigates mobility and contention issues by proposing a protocol named TCP Sintok. This protocol comprises two mechanisms: Delay-based Loss Detection Mechanism (LDM), and Contention Avoidance Mechanism (CAM). LDM was introduced to determine the cause of the packet loss by monitoring the trend of end-to-end delay samples. CAM was developed to adapt the sending rate (congestion window) according to the current network condition. A series of experimental studies were conducted to validate the effectiveness of TCP Sintok in identifying the cause of packet loss and adapting the sending rate appropriately. Two variants of TCP protocol known as TCP NewReno and ADTCP were chosen to evaluate the performance of TCP Sintok through simulation. The results demonstrate that TCP Sintok improves jitter, delay and throughput as compared to the two variants. The findings have significant implication in providing reliable data transfer within MANET and supporting its deployment on mobile device communication

Greediness control algorithm for multimedia streaming in wireless local area networks

This work investigates the interaction between the application and transport layers while streaming multimedia in a residential Wireless Local Area Network (WLAN). Inconsistencies have been identified that can have a severe impact on the Quality of Experience (QoE) experienced by end users. This problem arises as a result of the streaming processes reliance on rate adaptation engines based on congestion avoidance mechanisms, that try to obtain as much bandwidth as possible from the limited network resources. These upper transport layer mechanisms have no knowledge of the media which they are carrying and as a result treat all traffic equally. This lack of knowledge of the media carried and the characteristics of the target devices results in fair bandwidth distribution at the transport layer but creates unfairness at the application layer. This unfairness mostly affects user perceived quality when streaming high quality multimedia. Essentially, bandwidth that is distributed fairly between competing video streams at the transport layer results in unfair application layer video quality distribution. Therefore, there is a need to allow application layer streaming solutions, tune the aggressiveness of transport layer congestion control mechanisms, in order to create application layer QoE fairness between competing media streams, by taking their device characteristics into account. This thesis proposes the Greediness Control Algorithm (GCA), an upper transport layer mechanism that eliminates quality inconsistencies caused by rate / congestion control mechanisms while streaming multimedia in wireless networks. GCA extends an existing solution (i.e. TCP Friendly Rate Control (TFRC)) by introducing two parameters that allow the streaming application to tune the aggressiveness of the rate estimation and as a result, introduce fair distribution of quality at the application layer. The thesis shows that this rate adaptation technique, combined with a scalable video format allows increased overall system QoE. Extensive simulation analysis demonstrate that this form of rate adaptation increases the overall user QoE achieved via a number of devices operating within the same home WLAN

エッジクラウドコンピューティングにおける資源割当手法に関する研究

Edge and cloud computing is expected as a novel network architecture for supporting people’s daily lives and work. In edge and cloud computing, users expect to receive a wide variety of services with different requirements in terms of transmission rates on various locations, so there are some important issues regarding efficient service provision to meet user requirements independently of their locations: (1) fair service provision in cloud computing, (2) service provision according to user requirements in edge computing, and (3) efficient transmission in edge and cloud computing. In this study, I propose efficient schemes to allocate and use network resources among flows to solve these issues in edge and cloud computing and show the effectiveness of the proposed schemes through simulation evaluations.北九州市立大

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

MANETs: Internet Connectivity and Transport Protocols

A Mobile Ad hoc Network (MANET) is a collection of mobile nodes connected together over a wireless medium, which self-organize into an autonomous multi-hop wireless network. This kind of networks allows people and devices to seamlessly internetwork in areas with no pre-existing communication infrastructure, e.g., disaster recovery environments. Ad hoc networking is not a new concept, having been around in various forms for over 20 years. However, in the past only tactical networks followed the ad hoc networking paradigm. Recently, the introduction of new technologies such as IEEE 802.11, are moved the application field of MANETs to a more commercial field. These evolutions have been generating a renewed and growing interest in the research and development of MANETs. It is widely recognized that a prerequisite for the commercial penetration of the ad hoc networking technologies is the integration with existing wired/wireless infrastructure-based networks to provide an easy and transparent access to the Internet and its services. However, most of the existing solutions for enabling the interconnection between MANETs and the Internet are based on complex and inefficient mechanisms, as Mobile-IP and IP tunnelling. This thesis describes an alternative approach to build multi-hop and heterogeneous proactive ad hoc networks, which can be used as flexible and low-cost extensions of traditional wired LANs. The proposed architecture provides transparent global Internet connectivity and address autocofiguration capabilities to mobile nodes without requiring configuration changes in the pre-existing wired LAN, and relying on basic layer-2 functionalities. This thesis also includes an experimental evaluation of the proposed architecture and a comparison between this architecture with a well-known alternative NAT-based solution. The experimental outcomes confirm that the proposed technique ensures higher per-connection throughputs than the NAT-based solution. This thesis also examines the problems encountered by TCP over multi-hop ad hoc networks. Research on efficient transport protocols for ad hoc networks is one of the most active topics in the MANET community. Such a great interest is basically motivated by numerous observations showing that, in general, TCP is not able to efficiently deal with the unstable and very dynamic environment provided by multi-hop ad hoc networks. This is because some assumptions, in TCP design, are clearly inspired by the characteristics of wired networks dominant at the time when it was conceived. More specifically, TCP implicitly assumes that packet loss is almost always due to congestion phenomena causing buffer overflows at intermediate routers. Furthermore, it also assumes that nodes are static (i.e., they do not change their position over time). Unfortunately, these assumptions do not hold in MANETs, since in this kind of networks packet losses due to interference and link-layer contentions are largely predominant, and nodes may be mobile. The typical approach to solve these problems is patching TCP to fix its inefficiencies while preserving compatibility with the original protocol. This thesis explores a different approach. Specifically, this thesis presents a new transport protocol (TPA) designed from scratch, and address TCP interoperability at a late design stage. In this way, TPA can include all desired features in a neat and coherent way. This thesis also includes an experimental, as well as, a simulative evaluation of TPA, and a comparison between TCP and TPA performance (in terms of throughput, number of unnecessary transmissions and fairness). The presented analysis considers several of possible configurations of the protocols parameters, different routing protocols, and various networking scenarios. In all the cases taken into consideration TPA significantly outperforms TCP

Improving TCP performance over heterogeneous networks : The investigation and design of End to End techniques for improving TCP performance for transmission errors over heterogeneous data networks.

Transmission Control Protocol (TCP) is considered one of the most important protocols in the Internet. An important mechanism in TCP is the congestion control mechanism which controls TCP sending rate and makes TCP react to congestion signals. Nowadays in heterogeneous networks, TCP may work in networks with some links that have lossy nature (wireless networks for example). TCP treats all packet loss as if they were due to congestion. Consequently, when used in networks that have lossy links, TCP reduces sending rate aggressively when there are transmission (non-congestion) errors in an uncongested network. One solution to the problem is to discriminate between errors; to deal with congestion errors by reducing TCP sending rate and use other actions for transmission errors. In this work we investigate the problem and propose a solution using an end-to-end error discriminator. The error discriminator will improve the current congestion window mechanism in TCP and decide when to cut and how much to cut the congestion window. We have identified three areas where TCP interacts with drops: congestion window update mechanism, retransmission mechanism and timeout mechanism. All of these mechanisms are part of the TCP congestion control mechanism. We propose changes to each of these mechanisms in order to allow TCP to cope with transmission errors. We propose a new TCP congestion window action (CWA) for transmission errors by delaying the window cut decision until TCP receives all duplicate acknowledgments for a given window of data (packets in flight). This will give TCP a clear image about the number of drops from this window. The congestion window size is then reduced only by number of dropped packets. Also, we propose a safety mechanism to prevent this algorithm from causing congestion to the network by using an extra congestion window threshold (tthresh) in order to save the safe area where there are no drops of any kind. The second algorithm is a new retransmission action to deal with multiple drops from the same window. This multiple drops action (MDA) will prevent TCP from falling into consecutive timeout events by resending all dropped packets from the same window. A third algorithm is used to calculate a new back-off policy for TCP retransmission timeout based on the network¿s available bandwidth. This new retransmission timeout action (RTA) helps relating the length of the timeout event with current network conditions, especially with heavy transmission error rates. The three algorithms have been combined and incorporated into a delay based error discriminator. The improvement of the new algorithm is measured along with the impact on the network in terms of congestion drop rate, end-to-end delay, average queue size and fairness of sharing the bottleneck bandwidth. The results show that the proposed error discriminator along with the new actions toward transmission errors has increased the performance of TCP. At the same time it has reduced the load on the network compared to existing error discriminators. Also, the proposed error discriminator has managed to deliver excellent fairness values for sharing the bottleneck bandwidth. Finally improvements to the basic error discriminator have been proposed by using the multiple drops action (MDA) for both transmission and congestion errors. The results showed improvements in the performance as well as decreases in the congestion loss rates when compared to a similar error discriminator.Ministry of Higher Education and King Saud University in Saudi Arabia

Improving TCP Performance in the Mobile, High Speed, Heterogeneous and Evolving Internet

Ph.DDOCTOR OF PHILOSOPH