TCP Retransmission Timeout Algorithm Using Weighted Medians

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.

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.

Recent years internet blended multi-hop wireless networking technology experience growth in diverse sectors ranging from health care to the defense industry. An immense internet data traffic quantity depends on the transmission control protocol (TCP) due to its reliable data carrying ability between end process. TCP's congestion control algorithm performance under multi-hop wireless networks (MWN) is heavily influenced by retransmission timer (RT timer). The retransmission timeout (RTO) algorithm governs RT timer's clock duration. In MWN environment, Jacobson RTO prediction fails to endure sudden traffic delay spike that occurs due to inadequate recovery time during a link failure. In this paper, an analysis carried out to study the RTO's negative impact on TCP's performance under the multi-hop wireless scenario.

TCP transmission is mostly governed by Re–Transmission Timeout (RTO) values of each segment sent across the network. It is found that the fairness in the RTO is more need on effective management of the TCP. Most of the model calculates the RTO based on the Round Trip Time (RTT), which is the critical parameter in determining the behaviors of TCP. The RTO lacks in reflecting network foot prints. This paper presents algorithm which improves RTO, considering the network foot print for each segment sent across the network. Algorithm uses 3–way connection establishments RTT of TCP as the primary value to fine-tune the RTO. Using this RTT as base value and Advertise Window value of the receiver it calculates the rate of change of throughput as function of time to fix the RTO. The experiment has being simulated through OPNET.

The transmission control protocol (TCP) was designed to provide reliable end-to-end delivery of data over unreliable networks. Most TCP implementations have been carefully optimised to operate in wired networks. Deploying standard TCP in wireless mobile ad hoc networks (MANETs) can lead to poor performance of TCP. The main problem that faces TCP in ad hoc networks is its inability to distinguish between packet losses due to congestion and those due to route failures. In order to adapt TCP to MANET environment, several TCP improvements that differentiate between different types of packet losses using explicit notification messages from lower layers have been proposed. In this paper, we propose a new end-to-end technique that does not require any participation from intermediate nodes within the network. The proposed scheme relies on expanding the retransmission timeout (RTO) interval in order to reduce the number of timeouts that do not result from congestion. Simulation results show that our scheme can improve TCP throughput and delay. It outperformed the standard TCP in detailed simulation experiments.

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.

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.

this paper focuses on the effects of the TCP Initial Retransmission Timeout (RTO) and Round Trip Time (RTT) on one hand and the dictionary attack on the other hand. A dictionary attack model of two servers has been programmed and applied using TCL language. The delay time to find the correct password in the dictionary has been assigned to NS2 simulation network. The value of the initial RTO has been set to 0.5, 1, 3, 5, 10 seconds, with Round Trip Times (RTT) of 1, 5, 10, 20, and 50 milliseconds in the simulated network to study different situations. The results obtained were found to be consistent with previous findings associated to initial RTO values between 0.5s to Is for network performance. The additional values introduced here are for network security purposes as they represent the best values for which the attacker finds difficulty to accomplish dictionary attack.

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.

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.

In Long Term Evolution (LTE) systems both radio and transport congestion might occur due to the increasing data traffic generated by smart devices accessing popular Internet applications. Packets arriving at a congested resource experience increased delay, random discards and eventually tail drops as a consequence of network side buffer overflows. The LTE system delegates congestion management to the end-to-end protocols such as the Transmission Control Protocol (TCP), which is not an efficient solution in mobile systems. Reduced TCP transmission rate and significant degradations such as retransmission timeouts (RTO) are commonly present upon congestion. Therefore, instead of relying only on the congestion management capabilities of the TCP, efficient network side mechanisms are needed to achieve harmonic system operation and prevent unnecessary TCP degradations. This was recognized by 3GPP resulting in the specification of the User Plane Congestion management (UPCON) framework. Nevertheless, the real time congestion detection and management mechanisms are not in the scope of UPCON. As the prerequisite of any efficient congestion management technique is the capability of accurate congestion detection, this paper provides a survey of the existing solutions, analyses their shortcomings and proposes a congestion detection method that correlates the delay skewness and the pattern of burst losses collected by user traffic monitoring. The method is presented via measurements from an LTE lab network.

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.

Transmission Control Protocol (TCP) used multipath for achievement of data instantaneously to improve its performance.Anyhow, the preceding TCP protocol in the Internet of Things (IoT) network experienced a struggle to transmit a superior number of subflows.The Bottleneck Bandwidth and Round-trip propagation time (BBR) generate a network path model by assessing the available bottleneck bandwidth and the minimal Round-Trip Time (RTT) to achieve the desired delivery rate, consequently reducing the latency.However, some research studies have indicated the presence of significant fairness issues related to RTT in the BBR algorithm.To overcome this issue, we proposed a New Enhanced Congestion Control (CC) mechanism referred to as Bottleneck Bandwidth and Round-trip propagation time (BBR-NEnh) which is cross-layer-based efficient congestion control in a 5G IoT heterogeneous network.The proposed work encloses three sequential processes including grid-based network construction, improved two-factor clustering, and Hybrid TCP congestion control.Initially, we perform network construction by grid-based to improve the information transfer rate and increase connectivity among devices.After that, IoT devices are clustered based on two factors congestion and buffer space using the Improved K-Medoids algorithm (IK-Med) by the edge server.Further, to enhance the efficiency of TCP, the proposed research adopts a BBR-NEnh by optimizing the congestion control parameters using the Adaptive Tunicate Swarm Optimization (ATSO) algorithm.Finally, the dynamic handover and adjustment of windows are performed based on the Fast Adaptation Technique (FAdT) which can send the data quickly without congestion.The implementation of the proposed research is carried out using NS-3.26 and the performance of the proposed BBR-NEnh model is evaluated using a variety of performance metrics, including goodput, delay, packet loss, queue length, transmission rate, and throughput.The performance of the proposed work is compared with IMPRTT (Improved RTT) and CW-IoT (Congestion Window Algorithm for the Internet of Things).The average throughput, goodput, and packet loss of the proposed work for 1500 packets size are 4, 0.9, and 10.3, and the average delay, transmission rate, and queue length over 150 seconds are 0.004, 2500, and 36.1, demonstrating that the proposed work outperforms earlier work like IMPRTT and CW-IoT.