RSW Algorithm for TCP Dynamic Retransmission Timeout

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.

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.

The increased number and diversity of underlying networks have made transparent network migration a necessity. Through its support for multi-homing the Stream Control Transmission Protocol (SCTP) enables seamless network mobility by abstracting multiple underlying physical paths into a single end-to-end association. One of these paths is selected as the primary. When a number of retransmission failures occur on the primary path, switchover is initiated to a secondary path. The number of retransmission attempts before switchover is initiated can be configured; however, the delay between each retransmission is managed internally in SCTP using a Retransmission TimeOut (RTO) value. This paper shows that the current SCTP mechanism for calculating RTO values is inappropriate in WLAN environments, since increased Round Trip Times (RTT) significantly distort RTO calculations. Experimental and simulated results indicate that SCTP behaves in a counterintuitive manner which allows more time for switchover as network conditions degrade: delays of up to 187 seconds can be experienced before switchover occurs. We show that additional SCTP parameters need to be carefully configured in order to reduce this switchover delay to a more acceptable level.

This letter presents a new retransmission timeout (RTO) algorithm based on recursive weighted median (RWM) filters for the transmission control protocol (TCP). The RTO algorithm utilized in current TCP implementations is Jacobson's algorithm, which is based on recursive linear filtering. While linear filters are adequate for estimation in Gaussian signal environments, the round trip time (RTT) signals filtered to determine the RTOs are often impulsive. Thus, Jacobson's algorithm is not effective in many cases. The proposed algorithm employs RWM filters that yield improved performance when operating on RTT signals with heavy tailed statistics. Simulation results show that the proposed method yields significantly tighter RTT bounds than Jacobson's method over Internet traffic with heavy tailed statistics.

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.

In this paper, we address the problem of QoS (Quality of Service) provisioning for multi-user video streaming over multiple heterogeneous wireless networks based on the distributed, cross-layer design framework. By jointly considering the rate allocation and the Joint Source-Channel Coding (JSCC), our proposal aims at maximizing the QoS provisioning under the given resource constraint. At first, we develop and evaluate a framework for optimal video rate allocation over multiple networks based on the observed Available Bit Rate (ABR) and the Round Trip Time (RTT) over each access network, as well as the video rate-distortion characteristics. The rate allocation is formulated as a convex optimization problem that minimizes the sum of all video streams expected distortion. Then, we propose an analytical JSCC scheme for error-resilient scalable encoded video, and integrate the JSCC with the specific rate allocation algorithm to improve the constructed video quality by optimally applying the appropriate channel coding rate given the constraints imposed by the transmission rate and the prevailing channel conditions. Objective and subjective simulation results are provided which demonstrate the effectiveness of our proposed joint scheme.

This paper provides a preliminary investigation into the impact a link with changing propagation delay has on the performance of TCP file transfers. We investigate over a dozen different variable delay patterns, based on spacecraft movement. We highlight the performance impact of such variability, paying close attention to TCP's retransmission timer, which is based on the observed round-trip time of the network path. In addition, we explore one scenario in which the round-trip time across a network path suddenly changes due to a large change in the path between the two end-points. We conclude that the variable delay network paths studied in this paper do not drastically impact TCP performance.

The General Packet Radio Service (GPRS) is being deployed by GSM network operators world-wide, and promises to provide users with “always-on” data access at bandwidths comparable to that of conventional fixed-wire telephone modems. However, many users have found the reality to be rather different, experiencing very disappointing performance when, for example, browsing the web over GPRS. In this paper, we examine the causes, and show how unfortunate interactions between the GPRS link characteristics and TCP/IP protocols lead to poor performance. A performance characterization of the GPRS link-layer is presented, determined through extensive measurements taken over production networks. We present measurements of packet loss rates, bandwidth availability, link stability, and roundtrip time. The effect these characteristics have on TCP behavior are examined, demonstrating how they can result in poor link utilization, excessive packet queueing, and slow recovery from packet losses. Further, we show that the HTTP protocol can compound these issues, leading to dire WWW performance. We go on to show how the use of a transparent proxy interposed near the wired-wireless border can be used to alleviate many of these performance issues without requiring changes to either client or server end systems.

This paper presents a trace-driven simulation study of two classes of retransmission timeout (RTO) estimators in the context of real-time streaming over the Internet. We explore the viability of employing retransmission timeouts in NACK-based (i.e., rate-based) streaming applications to support multiple retransmission attempts per lost packet. The first part of our simulation is based on trace data collected during a number of real-time streaming tests between dialup clients in all 50 states in the U.S. (including 653 major U.S. cities) and a backbone video server. The second part of the study is based on streaming tests over DSL and ISDN access links. First, we define a generic performance measure for assessing the accuracy of hypothetical RTO estimators based on the samples of the round-trip delay (RTT) recorded in the trace data. Second, using this performance measure, we evaluate the class of TCP-like estimators and find the optimal estimator given our performance measure. Third, we introduce a new class of estimators based on delay jitter and show that they significantly outperform TCP-like estimators in NACK-based applications with low-frequency RTT sampling. Finally, we show that high-frequency sampling of the RTT completely changes the situation and makes the class of TCP-like estimators as accurate as the class of delay-jitter estimators.

The Constrained Application Protocol (CoAP) enables RESTful communication for resource-constrained IoT devices over UDP. However, in Separate Response mode, delayed or lost Empty Acknowledgments (ACKs) can trigger unnecessary retransmissions, causing duplicate responses, energy waste, and higher latency. This paper proposes a cooperative retransmission control scheme that enhances CoAP reliability without modifying its standard. The client employs an adaptive retransmission timeout (RTO) using an exponentially weighted moving average (EWMA) of round-trip times, while the server applies a twothreshold duplicate suppression mechanism that differentiates between transient delays and actual losses. Together, these components dynamically balance reliability and responsiveness. Simulation and analytical results in 6LoWPAN environments show reductions of up to 45 % in redundant messages, 25 % in latency, and 18 % in energy consumption compared with baseline CoAP. The proposed scheme provides a lightweight, fully compliant improvement for scalable, low-power IoT networking.

Due to the dynamic nature of the Internet, the characteristic parameters of network paths are continually changing. The round trip time (RTT) can be used to estimate retransmission timeouts with reasonable accuracy. However, using RTT to evaluate forward or backward delays is not suitable. By identifying this shortcoming, we propose a relative delay estimator (RDE) to make a distinction between all available paths and build a retransmission policy based on it.

A vertical handoff occurs when a multi-homed mobile host roams from a network to another heterogeneous network. Nevertheless, the network before and after a vertical handoff usually have drastic different characteristics. In this paper, we proposed an end-to-end based Vertical-handoff Aware TCP, called VA-TCP, which promptly adapts TCP behavior during a vertical handoff. When entering a new network, on the basis of packet-pair algorithm, VA-TCP dynamically estimates the new network bandwidth and round-trip time by using out-of-band ICMP messages. After that, by the estimated bandwidth and round-trip time, VA-TCP adjusts the TCP parameters including congestion window size, slow-start threshold, retransmission timer and RTTs to respond to the new network environment. From the experimental result, the TCP throughput by the proposed VA-TCP scheme is around 2 times that by the original TCP.

Changes involving the health care economic landscape have affected physicians' workflow, productivity, compensation structures, and culture. Ongoing Federal legislation regarding regulatory documentation and imminent payment-changing methodologies have encouraged physician consolidation into larger practices, creating affiliations with hospitals, multidisciplinary medical specialties, and integrated delivery networks. As subspecialization and evolution of care models have accelerated, independent medical groups have broadened ancillary service lines by investing in enterprises that compete with hospital-based (academic and nonacademic) entities, as well as non-physician- owned multispecialty enterprises, for both outpatient and inpatient services. The looming and dramatic shift from volume- to value-based health care compensation will assuredly affect urology group compensation arrangements and productivity formulae. For groups that can implement change rapidly, efficiently, and harmoniously, there will be opportunities to achieve the Triple Aim goals of the Patient Protection and Affordable Care Act, while maintaining a successful medical-financial practice. In summary, implementing new payment algorithms alongside comprehensive care coordination will assist urology groups in addressing the health economic cost and quality challenges that have been historically encountered with fee-for-service systems. Urology group leadership and stakeholders will need to adjust internal processes, methods of care coordination, cultural dependency, and organizational structures in order to create better systems of care and management. In response, ancillary services and patient throughput will need to evolve in order to adequately align quality measurement and reporting systems across provider footprints and patient populations.

Full-feedback contention window adaption for IEEE 802.11 WLANs

Abstract— Wireless local area networks is used by Sermo’s GNSS-CORS to renew its reference position from Department of Engineering Universitas Gadjah Mada’s CORS. The problem is, that wireless local area network should meet the bandwidth requirement and the operation should not break the frequency usage regulation in Indonesia. This reseach was conducted to study the bandwidth needs of Sermo’s dam deformation monitoring system, to design the topology, and calculating EIRP of all wireless network link in the system that meet the regulations of the Ministry of Communications and Information Republik of Indonesia. Wireless local area networks was built to connect Sermo, Faculty of Engineering Universitas Gadjah Mada and Balai Besar Wilayah Sungai Serayu Opak’s office. Design implementation and testing of broadband wireless equipment at unlicence frequency of 2.4 GHz (Motorola PTP 100) and 5.8 GHz (Motorola Motorola PTP and PTMP 5750AP/5760SM 58 300) resulted in 2.22 Mbps bandwidth that is much greater than that required, 128.7 kbps, and the average round trip delay of 19.2 ms. Antenna alignment information can be used as a reference for wireless communications equipment maintenance.