Congestion Window Growth Research Articles

Performance evaluation of the congestion severity aware rate regulation (CSRR) algorithm in wireless body area networks

SummaryWireless body area network (WBAN) is a potential low‐cost technology for privacy‐sensitive telemedicine and e‐health monitoring and services. However, it faces limited protocol and physical resource support challenges, which can result in packet transfer difficulties. In particular, WBAN requires an emergency‐aware technology that ensures a promising quality of service (QoS). One significant issue affecting QoS and energy efficiency in WBAN is congestion. Effective congestion control techniques are essential for achieving proper load balancing. To address these challenges, we propose a congestion severity aware rate control (CSRR) algorithm that enhances packet transmission rate by reducing packet losses and retransmissions. The CSRR algorithm incorporates a fuzzy controller to predict congestion rates based on runtime metrics. To regulate congestion window growth in different algorithm phases, we introduce sequences such as the Fibonacci retracement sequence, knight's move sequence, and the binary logarithm of the primorial sequence to regulate congestion window growth in the different phases of the proposed algorithm. We mathematically analyze the proposed CSRR algorithm using a Markov model. The simulation results demonstrate the superiority of our algorithm compared to existing approaches. Specifically, our algorithm achieves significant optimizations in terms of throughput (52.92%), packet loss (38.11%), delay (37.23%), and remaining energy (36.86%) when compared to existing algorithms.

TCCC: A Throughput Consistency Congestion Control Algorithm for MPTCP in Mixed Transmission of Long and Short Flows

Existing congestion control algorithms for MPTCP that care about only long flow transmission aim at the Congestion-Avoidance (CA) phase and they need a long time to reach convergence states. We verified that the exponential growth of congestion window (cwnd) in the uncoupled Slow-Start (SS) leads to not only unfairness to TCP but also buffer overflow due to burst data. Moreover, these algorithms cannot support fair bandwidth sharing among TCP/MPTCP flows before reaching convergence at the bottleneck, which may reduce the transmission efficiency of short flows and even hurts long flows. In this paper, we propose a Throughput Consistency Congestion Control (TCCC) algorithm consisting of Coupled Slow-Start (CSS) and Aggressive Congestion Avoidance (ACA). To prevent packet loss caused by excessive burst data, CSS couples the increment of subflows’ cwnd and reset the ssthresh value to safely move the flows to CA when it achieves expected throughput. Based on CSS, ACA periodically detects path states and allocates the same throughput increment as the best TCP to subflows to achieve fair bandwidth share in CA. Finally, we implement TCCC in both NS3 and real testbed. The results show that TCCC reduces retransmissions, improves transmission efficiency, and maintains better fairness.

Path-Rank-Based Data Chunk Scheduling for Concurrent Multipath Transmission

Abstract The device equipped with a multi-homing feature optimally exploits multiple network interfaces in modern communications networks such as the Internet of Things (IoT) and machine-to-machine communication using the concurrent multipath transfer (CMT). This enhances system performance by concurrently scheduling data chunks on multiple network paths. For a while, several scheduling criteria have been developed to optimize performance. However, it has been identified that CMT still suffers from many serious problems, such as spurious retransmission, receiver buffer blocking, improper congestion window (CWND) growth, re-ordering and long round trip time, resulting in poor performance. These problems occur due to the asymmetric nature of path characteristics. Thus, this paper introduces a path rank-based CMT (R-CMT) that schedules data chunks according to the rank of the path. The proposed scheduling method calculates the rank of each network path based on the ratio of successfully received and transmitted chunks. The simulation results indicate that the proposed R-CMT scheduling achieves higher performance in terms of network latency, throughput and CWND growth.

NexGen D-TCP: Next Generation Dynamic TCP Congestion Control Algorithm

With the advancement of wireless access networks and mmWave New Radio (NR), new applications emerged, which requires a high data rate. The random packet loss due to mobility and channel conditions in a wireless network is not negligible, which degrades the significant performance of the Transmission Control Protocol (TCP). The TCP has been extensively deployed for congestion control in the communication network during the last two decades. Different variants are proposed to improve the performance of TCP in various scenarios, specifically in lossy and high bandwidth-delay product (high-BDP) networks. Implementing a new TCP congestion control algorithm whose performance is applicable over a broad range of network conditions is still a challenge. In this article, we introduce and analyze a Dynamic TCP (D-TCP) congestion control algorithm over mmWave NR and LTE-A networks. The proposed D-TCP algorithm copes up with the mmWave channel fluctuations by estimating the available channel bandwidth. The estimated bandwidth is used to derive the congestion control factor N. The congestion window is increased/decreased adaptively based on the calculated congestion control factor. We evaluated the performance of D-TCP in terms of congestion window growth, goodput, fairness and compared it with legacy and existing TCP algorithms. We performed simulations of mmWave NR during LOS NLOS transitions and showed that D-TCP curtails the impact of under-utilization during mobility. The simulation results and live air experiment points out that D-TCP achieves 32.9% gain in goodput as compared to TCP-Reno and attains 118.9% gain in throughput as compared to TCP-Cubic.

Tuning the Aggressive Slow-Start Behavior of MPTCP for Short Flows

With the widespread availability of multi-homed devices, enabling multiple paths by utilizing multipath TCP (MPTCP) in current networks is a common practice to improve the performance and robustness. Although MPTCP improves both bandwidth efficiency and network reliability, the transmission performance for short flows can become worse than regular single path TCP when the concurrent subflows transferred through a shared bottleneck due to MPTCP’s aggressive slow start behavior, which is uncoupled, and each subflow behaves independently as regular TCP, affecting MPTCP and concurrent traffic at the bottleneck. In this paper, we first reveal that the MPTCP’s aggressive behavior in slow start causes timeouts and throughput collapse. We further present the design and implementation of GSAM, which employs the theoretical analysis to derive the appropriate threshold for smoothing the congestion window growth in GSAM according to the network conditions in slow start phase and leverages congestion detection and control at end-host to avoid buffer overflow under concurrent MPTCP connections with multiple subflows sharing the bottleneck. The experimental results based on the real implementations show that the GSAM reduces the completion time by up to 80% while retaining high Goodput for large flows as MPTCP.

End-to-End Loss Based TCP Congestion Control Mechanism as a Secured Communication Technology for Smart Healthcare Enterprises

Many smart healthcare centers are deploying long distance, high bandwidth networks in their computer network infrastructure and operation. Transmission control protocol (TCP) is responsible for reliable and secure communication of data in these medial infrastructure networks. TCP is reliable and secure due to its congestion control mechanism, which is responsible for detecting and reacting to the congestion in the network. Many TCP congestion control mechanisms have been developed previously for different operating systems. TCP CUBIC, TCP Compound, and TCP Fusion are the default congestion control mechanism in Linux, Microsoft Windows, and Sun Solaris operating systems, respectively. The earliest congestion control mechanism Standard TCP acts as the trademark congestion control mechanism. The exponential growth of congestion window ( $cwnd$ ) in slow start phase of the TCP CUBIC causes burst losses of packets, and TCP flows did not share available link bandwidth fairly. The prime aim of this paper is to enhance the performance of TCP CUBIC for long distance, high bandwidth secured networks to achieve better performance in medical infrastructure, concerning packet loss rate, protocol fairness, and convergence time. In this paper, congestion control module for slow start is proposed, which reduces the effect of the exponential growth of $cwnd$ by designing the new limits of $cwnd$ size in slow start phase, which in turn decreases the packet loss rate in healthcare networks. NS-2 is used to simulate the experiments of enhanced TCP CUBIC and state-of-the-art congestion control mechanisms. Results show that the performance of enhanced TCP CUBIC outperforms by 18% as compared with the state-of-the-art congestion control mechanisms.

Achieving Robust Mobile Web Content Delivery Performance Based on Multiple Coordinated QUIC Connections

In order to minimize the downloading time of short-lived applications like Web browsing, Web application, and short video clips, the recently standardized HTTP/2 adopts stream multiplexing on one single TCP connection. However, aggregating all content objects within one single connection suffers from the head-of-line blocking issue. Quick UDP Internet connection (QUIC), by eliminating such an issue on the basis of UDP, is expected to further reduce the content downloading time. However, in mobile network environments, the single connection strategy still leads to a degraded and high-variant completion time due to the unexpected hindrance of congestion window growth caused by the common but uncertain fluctuations in round trip time and also random loss event at the air interface. To retain resilient congestion window against such network fluctuations, we propose an intelligent connection management scheme based on QUIC which not only employs adaptively multiple connections but also conducts a tailored state and congestion window synchronization between these parallel connections upon the detection of network fluctuation events. According to the performance evaluation results obtained from an LTE-A/Wi-Fi testing network, the proposed multiple QUIC scheme can effectively overcome the limitations of different congestion control algorithms (e.g., the loss-based new reno/CUBIC and the rate-based BBR), achieving substantial performance improvement in both median (up to 59.1%) and 95th completion time (up to 72.3%). The significance of this piece of work is to achieve highly robust short-lived content downloading performance against various uncertainties of network conditions as well as with different congestion control schemes.

Randomizing TCP payload size for TCP fairness in data center networks

As many-to-one traffic patterns prevail in data center networks, TCP flows often suffer from severe unfairness in sharing bottleneck bandwidth, which is known as the TCP outcast problem. The cause of the TCP outcast problem is the bursty packet losses by a drop-tail queue that triggers TCP timeouts and leads to decreasing the congestion window. This paper proposes TCPRand, a transport layer solution to TCP outcast. The main idea of TCPRand is the randomization of TCP payload size, which breaks synchronized packet arrivals between flows from different input ports. Based on the current congestion window size and the CUBIC’s congestion window growth function, TCPRand adaptively determines the proper level of randomness. With extensive ns-3 simulations and experiments, we show that TCPRand guarantees the superior enhancement of TCP fairness by reducing the TCP timeout period noticeably even in an environment where serious TCP outcast happens. TCPRand also minimizes the total goodput loss since its adaptive mechanism avoids unnecessary payload size randomization. Compared with DCTCP, TCPRand performs fairly well and only requires modification at the transport layer of the sender which makes its deployment relatively easier.

An adaptive data chunk scheduling for concurrent multipath transfer

Concurrent Multipath Transfer (CMT) is a transport layer protocol which provides concurrent data transfer over the multiple paths. CMT improves the available bandwidth utilization, fault tolerance, robustness and reliability of the network. However, in multipath data transfer, destination receives out-of-order data chunk due to dissimilar delay and bandwidth of each path. It causes the serious problem of receiver buffer blocking, unwanted congestion window (cwnd) reduction and unnecessary retransmission, which significantly degrades the performance of CMT. Thus, this paper proposes an adaptive data chunk scheduling for CMT (A-CMT). The proposed method uses path delay and bandwidth as a factor of data chunk scheduling to adapt path conditions. The simulation results show that proposed method achieves better performance in terms of throughput; file transfer time and congestion window growth. The proposed method improves average throughput up to 13%.

초고화질 스트리밍 서비스의 QoS를 향상시키기 위한 라우터 버퍼 기반의 혼잡 제어 기법

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

Adjusting the TCP Sending Rate and Retransmissions after Retransmission Timeouts Based on One-Way Queuing Delay in Wireless Mesh Network

Wireless Mesh Network (WMN) is regarded as a viable solution to provide broadband Internet access flexibly and cost efficiently. Improving the performance of Transmission Control Protocol (TCP) in WMNs is an active research area in the networking community. The existing solutions proposed for improving the TCP performance has concentrated on differentiating the DATA packet drops in the forward direction induced by both network congestion as well as transmission errors. However, the recent studies show that in WMNs packet drops occur not only in the forward direction but also in the reverse direction particularly due to hidden terminal, hidden capture terminal, link asymmetry etc. The loss of ACK packets in the reverse direction cause frequent retransmission timeouts subject to needless retransmissions and unnecessary slowing down the growth of congestion window, which causes the performance degradation of TCP. In this paper, we introduce a sender side TCP algorithm, called detection of packet loss (DPL), which is capable to distinguish the type of packet drops either DATA or ACKs caused by transmission errors as well as network congestion based on one-way queuing delay and react accordingly. To justify our contributions, we implement DPL in Qualnet simulator and compare its performance against existing TCP solutions via extensive simulations. Our simulation results show that the proposed algorithm can accurately distinguish the type of packet drops whether it is a DATA or ACK caused by transmission error or congestion and can significantly improve the performance under a wide range of scenarios in WMNs.

TCP congestion avoidance algorithm based on adaptive congestion window

Concerning the unsmooth growth of congestion window at congestion avoidance phase of TCP Reno,the traditional AIMD algorithm was researched and an improved congestion avoidance algorithm was proposed: a logarithmic function based on the growth of congestion window was adopted.In this new algorithm,additive factor is usually increased quickly when the network is doing well and less so when network situation is getting to congestion.The mathematical analysis shows the feasibility of the new algorithm,and its throughput,fairness and friendliness were evaluated by NS simulation.The simulation results show the effectiveness of the algorithm.

Performance Enhancement of DCCP TCP-like over Long Delay Link Networks

The performance of DCCP TCP-like degrades significantly over long delay link networks. Despite the TCP-like congestion control mechanism follows the TCP SACK, the performance is really affected by the congestion window growth algorithms as employed by Jacobson based TCP variants. In this paper, all the experiments are done using Network Simulator ns-2, and we manipulated the congestion window size drop during congestion avoidance phase to enhance the performance of DCCP TCP-like over long delay link networks. Instead of halving the current congestion window when congestion events are detected, the reduction of current congestion window drop has been shown to improve the DCCP TCP-like throughput with minimal drop packet percentage.

TCP Over the IEEE 802.15.3 MAC: Analysis and Simulation

High data rate wireless technologies are now becoming a reality, and have spurred the development of new applications that were previously hindered by the lack of capacity. In particular, it is now possible to stream high definition movies from a laptop to a sound system and flat screen television seamlessly. One of the key enablers of such applications is the IEEE 802.15.3 medium access control (MAC) protocol, which is designed to support bandwidth intensive applications in wireless personal area networks (WPANs). A key observation is that a significant number of multi-media applications rely on the transmission control protocol (TCP). Unfortunately, little works have conducted a thorough performance study of TCP over the IEEE 802.15.3 MAC. Moreover, the IEEE 802.15.3 specification does not specify any strategies for allocating time slots. This paper therefore contributes to the current state-of-the-art in the following manners. From our extensive analytical and simulation studies, we reveal the impacts of different channel time allocation methods and acknowledgment policies on the performance of TCP, with particular attention on round trip time, congestion window growth, and packet recovery. We then present the following guidance to application developers: (1) channel time allocations (CTAs) should be distributed evenly over the superframe and have durations determined by TCP’s maximum congestion window, (2) CTA positioning has no impact on TCP’s performance, and (3) the specified delayed acknowledgment policy needs to be augmented with an adaptive algorithm that adjusts its burst size dynamically to varying bit error rate (BER).

An analytic model of TCP performance over multi-hop wireless links with correlated channel fading

In this paper, we present an analytic model for TCP performance over wireless channels with highly correlated fading characteristics. The wireless channel TCP segment loss process is modeled using a Linear Algebraic Queueing Theory representation of a hidden Markov chain that can incorporate autocorrelations in successive segment losses. The segment loss model is then used in the development of a discrete time Markov chain representation of the evolution of the TCP congestion window. Variability in the round-trip time (RTT) distribution and sublinear congestion window growth is also incorporated into the model. We derive transient and steady-state performance measures such as the mean and variance of the congestion window size and throughput for various error autocorrelation and RTTs.

Measuring the evolution of transport protocols in the internet

In this paper we explore the evolution of both the Internet's most heavily used transport protocol, TCP, and the current network environment with respect to how the network's evolution ultimately impacts end-to-end protocols. The traditional end-to-end assumptions about the Internet are increasingly challenged by the introduction of intermediary network elements (middleboxes) that intentionally or unintentionally prevent or alter the behavior of end-to-end communications. This paper provides measurement results showing the impact of the current network environment on a number of traditional and proposed protocol mechanisms (e.g., Path MTU Discovery, Explicit Congestion Notification, etc.). In addition, we investigate the prevalence and correctness of implementations using proposed TCP algorithmic and protocol changes (e.g., selective acknowledgment-based loss recovery, congestion window growth based on byte counting, etc.). We present results of measurements taken using an active measurement framework to study web servers and a passive measurement survey of clients accessing information from our web server. We analyze our results to gain further understanding of the differences between the behavior of the Internet in theory versus the behavior we observed through measurements. In addition, these measurements can be used to guide the definition of more realistic Internet modeling scenarios. Finally, we present several lessons that will benefit others taking Internet measurements.