A Switched Rate-based Robust Queue Management for Wireless Access Networks

Classical Transmission Control Protocol (TCP) designs have never considered the identity of the competing transport protocol as useful information to TCP sources in congestion control mechanisms. When competing against a TCP flow on a bottleneck link, a User Datagram Protocol (UDP) flow can unfairly occupy the entire link bandwidth and suffocate all TCP flows on the link. If it were possible for a TCP source to know the type of transport protocol that deprives it of link access, perhaps it would be better for the TCP source to react in a way which prevents total starvation. In this paper, we use coefficient of variation and power spectral density of throughput traces to identify the presence of UDP transport protocols that compete against TCP flows on bottleneck links. Our results show clear traits that differentiate the presence of competing UDP flows from TCP flows independent of round-trip times variations. Signatures that we identified include an increase in coefficient of variation whenever a competing UDP flow joins the bottleneck link for the first time, noisy spectral density representation of a TCP flow when competing against a UDP flow in the bottleneck link, and a dominant frequency with outstanding power in the presence of TCP competition only. In addition, the results show that signatures for congestion caused by competing UDP flows are different from signatures due to congestion caused by competing TCP flows regardless of their round-trip times. The results in this paper present the first steps towards development of more ’intelligent’ congestion control algorithms with added capability of knowing the identity of aggressor protocols against TCP, and subsequently using this additional information for rate control.

Multiple access methods in a wireless network allow multiple nodes to share a set of available channels for data transmission. The nodes can either compete or cooperate with each other to access the channel(s) so that either an individual or a group objective can be achieved. Game theory, which is a mathematical tool developed to understand the interaction among rational entities, can be applied to model and to analyze individual or group behaviour of nodes for multiple access in wireless networks. Game theory also enables us to model the selfish/malicious behaviour of nodes, and subsequently design the punishment or defense mechanisms for robust multiple access in wireless networks. In addition, game models can provide distributed solutions to the multiple access problems, which are based on solid theoretical foundations. In this survey, we provide a comprehensive review of the game models (e.g., noncooperative/cooperative, static/dynamic, and complete/incomplete information) developed for different multiple access schemes (i.e., contention-free and contention-based random channel access) in wireless networks. We consider time-division multiple access (TDMA), frequency-division multiple access (FDMA), and code-division multiple access (CDMA), ALOHA, and carrier sense multiple access (CSMA)-based wireless networks. In addition, game models for multiple access in dynamic spectrum access-based cognitive radio networks are reviewed. The major findings from the game models used for these different access schemes are highlighted. To this end, several of the key open research directions are outlined.

Transmission Control Protocol (TCP) is well-known for its poor performance over wireless networks. The main reason is the unstable condition of wireless networks. Previous attempts to improve TCP performance have been centered on techniques such as queue management, congestion window adjustment, delayed acknowledgement, and etc., because lower layers are hidden in the layered Open Systems Interconnection (OSI) model. With cross layer techniques, more precise data can be retrieved from lower layers for network condition estimation. In this paper, we analyze and model the factors that affect the TCP throughput in the lower layers. Consequently, to solve the TCP optimization problem, we formulate TCP performance as a Markov Decision Process (MDP). An extensive set of simulation experiments has been performed on NS-2 in order to illustrate that TCP performance can be enhanced by cooperatively adjusting lower layer factors.

Transmission control protocol (TCP) provides reliable connection-oriented services between any two end systems on the Internet. With TCP congestion control algorithm, multiple TCP connections can share network and link resources simultaneously. These TCP congestion control mechanisms have been operating effectively in wired networks. However, performance of TCP connections degrades rapidly in wireless and lossy networks. To sustain the throughput performance of TCP connections in wireless networks, design modifications may be required accordingly in the TCP flow control algorithm, and potentially, in association with other protocols in other layers for proper adaptations. In this chapter, we explain the limitations of the latest TCP congestion control algorithm, and then review some popular designs for TCP connections to operate effectively in wireless mesh network infrastructure.

The differentiated services (DiffServ) is proposed to provide packet level service differentiations in a scalable manner. To provide an end-to-end service differentiation to users having a connection over multiple domains, as well as a flow marker, an intermediate marker is necessary at the edge routers, and it should not be operated at a flow level due to a scalability problem. Due to this operation requirement, the intermediate marker has a fairness problem among the transmission control protocol (TCP) flows since TCP flows have intrinsically unfair throughputs due to the TCP's congestion control algorithm. Moreover, it is very difficult to resolve this problem without individual flow state information such as round trip time (RTT) and sending rate of each flow. In this paper, to resolve this TCP fairness problem of an intermediate marker, we propose a fair scalable marker (FSM) as an intermediate marker which works with a source flow three color marker (sf-TCM) operating as a host source marker. The proposed fair scalable marker improves the fairness among the TCP flows with different RTTs without per-flow management. Through the simulations, we show that the FSM can improve TCP fairness as well as link utilization in multiple domain DiffServ networks.

Long distance high bandwidth networks are spanning several continents and many remote Healthcare centers are centralizing their data centers for economic reasons. For the best performance of their data centers, TCP (Transmission Control Protocol) performance and data security are the main critical issues in these network scenarios. TCP performance is directly related to its congestion control mechanism which is responsible for detecting and reacting to the overload traffic on the network. Data security is related to the security mechanism being used by sender and receiver nodes during communication. Linux users, which have rapidly increased in the last five years and most of the Healthcare data centers are being deployed on the Linux operating system, focus the researchers to work on Linux to enhance its performance and security accordingly. The Linux operating system uses TCP CUBIC as a congestion control mechanism with TCP during communication. TCP CUBIC became the default congestion control mechanism of Linux in 2006 after kernel 2.6.18. TCP CUBIC is fundamentally a loss based TCP congestion control mechanism and at each packet loss detection, it reduces its Congestion Window ( $cwnd$ c w n d ) size 20 percent instead of 50 percent as in trademark congestion control mechanism Standard TCP. The aim of this paper is to design a new security mechanism that will work with TCP CUBIC to achieve the maximum possible performance and security over the network link. In this paper, Network Simulator 2 (NS-2) is used to compare the performance of TCP CUBIC with state-of-the-art mechanisms in long and short Round Trip Time (RTT), high bandwidth network scenarios. Results show that when new security mechanism is used with TCP CUBIC, overall better performance in the form of protocol fairness, TCP friendliness, goodput, and convergence time is achieved over the network link.

REDUP: a packet loss recovery scheme for real-time audio streaming over wireless IP networks

Due to the development and evolution in Cognitive Radio Networks (CRNs), the performance of Transmission Control Protocol (TCP) over CRN has become a crucial factor. TCP ensures the reliable end-to-end delivery of data, which is very important for the quality of service (QoS) requirements. TCP performance is severely affected by channel switching and channel availability in CRNs. The frequent occurrence of such events is due to CRNs nature which unnecessarily reduces the transmission rate which results in degrading the overall performance. In this paper, the authors investigate the limitations of using TCP NewReno (i.e., standard TCP) over CRNs, and further propose a cross-layer-design protocol based on an interaction between transport layer and lower layers to improve TCP performance over CRNs. The design is called Cross-Layer Design for TCP (TCP-CLD). It primarily aims at improving the TCP performance over CRNs via controlling and adapting a few selected TCP parameters. Specifically, the authors propose freezing the data transmission and re-transmission timeout (RTO) of TCP during the time of channel switching and channel unavailability. In addition, the authors update the size of congestion window (cwnd) of TCP based on the bandwidth and buffer size of relay nodes. Moreover, in the case of channel switching, the authors update the Round Trip Time (RTT) of TCP based on the new route delay. Simulation experiments have been conducted using ns-2 network simulator tool whereas results show that the proposed design achieves a significant performance improvement compared with the standard TCP NewReno protocol and the other known protocol, namely, TCP-CRAHN.

Power Line Communications (PLC) is regarded as the promising home networking technology for smart grid application since the PLC technology enables data transmission via the existing power line. The cooperation between PLC and Wireless Local Area Network (WLAN) will be an effective hybrid solution for Home Area Networks (HANs). In this paper, we consider that Transmission Control Protocol (TCP) packets are transmitting in heterogenous communications network based on PLC and WLAN. The existing TCP congestion control has low effectiveness due to the error-prone power line channel. Therefore, we propose an adaptive rate control to improve TCP performance in the hybrid PLC/WLAN HANs. We exploit the communications mechanism of WLAN to improve the PLC link utilization. The methodology of Heuristic Dynamic Programming (HDP) is employed to obtain the optimal rate policy. The bottleneck link capacity is estimated by neural network training. The sending rate is adapted according to the long-term quality of service (QoS) of applications. Simulation results illustrate that the proposed rate control effectively filters out the negative impact from the instantaneous impulse noise in PLC channel and outperforms the existing TCP.

Fast transmission control protocol (TCP) was previously proposed for high capacity network environments with long delay, and TCP with Snoop performs better than conventional TCP enhancements in mobile wireless network environments. However, FAST TCP has limitations when used over a dynamic mobile wireless link with a high frame error ratio (FER) and frequent delay changes due to the variable rate. We propose an enhanced TCP acceleration algorithm at the TCP sender side which efficiently adapts to the maximum transmission rate of a mobile wireless link using the round trip time (RTT) and virtual receiver window (RWND) information. The proposed algorithm provides superior performance over mobile wireless network environments.

Universal Mobile Telecommunications System (UMTS) is a third-generation cellular network that enables high-speed mobile Internet access. This paper evaluates and compares the performance of two well-known versions of Transmission Control Protocol (TCP), namely, Vegas and Reno, in a UMTS environment. Bulk data transfer was considered in the simulation with varying radio channel conditions. We assume that data losses are only due to the radio channel. Simulation results show that the performance of Vegas is worse than Reno even though data losses incurred by the radio channel are completely recovered by the UMTS radio link control layer. This has led us to conduct a thorough investigation on the behavior of Vegas in order to identify the cause of performance degradation in Vegas. The poor performance of Vegas is attributed to the UMTS radio interface characteristics which resulted in large and highly variable TCP round-trip times. Vegas would interpret the round-trip time variation as a sign of congestion, and consequently, shrink its window size which reduces the transmission rate. Furthermore, a sudden increase in the instantaneous round-trip time can trigger spurious timeouts at the TCP sender using Vegas which performs unnecessary retransmissions. Spurious timeouts can lead to significant throughput reduction. Reno, on the other hand, does not show any abnormality and delivers the expected performance.

This paper studies the impact of variable transmission delays on the Transmission Control Protocol (TCP). Sudden delay variations, which are not uncommon in mobile networks, may degrade the performance since they may cause spurious TCP timeouts. The most important parameter in this context is the TCP retransmission timer. In this paper, we analyze TCP’s round-trip time estimation for bulk data traffic over wireless links. The main contribution is a new analytical model that accurately predicts the timeout duration from given network parameters. As a first result, the model shows that the round-trip time sampling rate has a significant impact on the timer characteristics. Therefore, the standardized estimation algorithm does not harmonize well with timestamp-based measurement. Second, we quantify the risk of spurious TCP timeouts triggered by changing round-trip times, in particular long off periods. We conclude that delay variations are only critical when they are on the order of seconds.

The estimation of Retransmission Timeout (RTO) in Transmission Control Protocol (TCP) affects the throughput of the transmission link. If the RTO is just a little larger than the Round Trip Time (RTT), retransmissions will occur too often, and this increases congestion in the transmission link [1]. If the RTO is much larger than the RTT, the response to retransmit when a packet is lost will be too slow, and this will decrease the throughput in the transmission link. Currently, Jacobson's Algorithm [2] for estimation of RTO is implemented in TCP. He uses an Exponential Weighted Moving Average (EWMA) filter to estimate RTT and then determines RTO. The EWMA filter is good if RTT follows a Gaussian distribution. In reality, traffic on the Internet is bursty and follows a heavy-tailed distribution. Median filter has been recognized as a useful filter due to its edge preserving characteristic for image processing applications, and it performs well for heavy-tailed distributions. Thus the median filter is efficient for removing impulsive noise [3], [4]. In this paper, we demonstrate the applicability of the median filter in our simulations for timeout estimation in the presence of bursty traffic flows.

The proliferation of mobile demands and increasingly multifarious services pose challenges to both the radio access networks (RANs) and the core network (CN). Unfortunately, the traditional networks can hardly meet the increasing requirements. Software-defined network (SDN) and network virtualization (NV) are considered as innovative paradigms to stress these problems. By extending SDN to wireless networks, software-defined wireless network (SDWN) consisting of both RANs and CN can be obtained. In this paper, the RANs and CN of SDWN are virtualized to virtual access points (VAPs) and network slices respectively based on NV technology. Considering the system scenario that a SDWN consists of overlapping RANs, an optimal joint VAP and network slice selection scheme is proposed. To characterize the complicated transmission performance of VAPs and network slices, the concepts of utility function and network calculus are introduced, and the joint utility of VAPs and network slices is formulated based on which the optimal VAP and network slice combination corresponding to the maximal joint utility can be selected. The simulation results demonstrate the efficiency of the proposed scheme.

Based on the assumption of transmission control protocol (TCP) that packetsare lost due to congestion, TCP’s congestion control algorithms suchas fast retransmit/recovery (FRR) and retransmission timeouts (RTO)unconditionally reduce the transmission rate for every packet loss. When TCPoperates in wireless networks, however, FRRs/RTOs are often triggeredregardless of congestion due to sudden delay and wireless transmissionerrors. The congestion irrelative FRRs/RTOs incur TCP’s misbehaviorsuch as blindly halving the transmission rate, unnecessarily retransmittingthe outstanding packets which may be in the bottleneck queue. Although manyprevious studies have been proposed to detect the congestion irrelativeFRRs/RTOs, they paid little attention on effectively adjusting thetransmission rate for the detected congestion irrelative FRRs/RTOs.In this article, we propose an enhanced TCP to dynamically adjust itstransmission rate according to network conditions. Our scheme adjusts thetransmission rate in proportion to the available bandwidth in order toquickly utilize the available bandwidth, and also re-adjusts it in inverseproportion to the loss rate in order to avoid burst losses and longgo-back-Nretransmissions. By doing so, our scheme hassignificant effects to avoid the performance degradation caused by thecongestion irrelative FRRs/RTOs. Throughout the extensive experiments, weevaluate our scheme and compare it with previous works in terms of goodput,fairness, and friendless under various network topologies. The results showthat our scheme significantly outperforms previous studies while itmaintains the fair and friendly behavior to other TCP connections.