Standard Transmission Control Protocol Research Articles

Introducing a New TCP Variant for UAV networks following comparative simulations

The Transmission Control Protocol (TCP) is a reliable, connection oriented, congestion control mechanism, currently utilized the majority of both wired and wireless networks at the transport layer. An important function of TCP is the network congestion control mechanism; it governs the packet transmission rate and us enables the protocol to respond to congestion signals. Considering the wide spectrum of requirements stemming from unique network and channel characteristics, there exist numerous variants of TCP. While is commonly utilized in applications requiring reliable and ordered reception of packets, the standard TCP congestion control mechanism demonstrates poor performance in high-mobility wireless networking scenarios, as high mobility implies unreliable radio links and consecutive re-transmissions. In this paper we performed a comparative analysis of a spectrum of TCP variants, with the ultimate goal of deriving best practices to support real-time, yet reliable communication in high-mobility scenarios, with a special focus on aerial mobile ad hoc networks composed of Unmanned Aerial Vehicles (UAVs). Following the findings of this simulation evaluation, we introduce a new TCP variant for Flying Ad hoc Networks (FANETs), named Swarm HTCP (S-HTCP), which is shown to outperform the other variants in such network conditions.

Congestion control in high‐speed networks using the probabilistic estimation approach

SummaryNowadays, the bulk of Internet traffic uses Transmission Control Protocol (TCP) for reliable transmission. But the standard TCP's performance is very poor in high‐speed networks (HSNs), and hence, the core gigabytes links are usually underutilization. This problem has roots in conservative nature of TCP, especially in its Additive Increase Multiplicative Decrease (AIMD) phase. In other words, since TCP cannot figure out precisely the congestion status of the network, it follows a conservative strategy to keep the network from overwhelming. We believe that precisely congestion estimation in the network can solve this problem by avoiding unnecessary conservation. To this end, this paper proposes an algorithm, which considers packet loss and delay information jointly and employs a probabilistic approach to accurately estimation of congestion status in the network. To examine the proposed scheme performance, extensive simulations have been performed in the NS‐2 environment. Simulation results reveal that the proposed algorithm has better performance than existing algorithms in terms of bottleneck utilization, stability, throughput, and fairness.

An SDN-based true end-to-end TCP for wireless LAN

Segment losses due to intermittent connectivity and mobility lead to sub optimal performance of the Transmission Control Protocol (TCP). This is due to the fact that segment loss is considered as a binary signal for triggering congestion control and retransmission mechanisms at the TCP sender. In wired networks, segments are dropped due to congestion at the routers and the strategy of taking missed acknowledgment as an implicit signal for congestion control performs well. However, in wireless networks, segment losses are primarily due to mobility and transmission errors. Unlike many previous efforts, this paper proposes the design and implementation of Software Defined Network (SDN) assisted TCP which does not require the wireless Access Points (APs) to be TCP-aware and preserves end to end semantics. Further, no changes are required to be done in TCP protocol implementation at the end-hosts. The proposed approach utilizes the programmability provided by the SDN paradigm to intelligently trigger the spurious timeout detection and response algorithms, already implemented in standard TCP. The proposed approach is compared with the standard TCP and SDN assisted Zero Window based approach on Linux kernels using virtual data-plane switches and APs provided by the Mininet-WiFi platform. The implementation results establish the applicability of the proposed approach.

TCP-CLD: Cross-Layer Design for Improving TCP Performance Over Cognitive Radio 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.

Multimodal Retransmission Timer for LPWAN

Low-power wide-area networks (LPWANs) are experiencing high momentum as an inexpensive solution for enabling the Internet-of-Things (IoT) applications. Recent Internet connectivity support developments are expected to further fuel the adoption of LPWAN. However, the latter present challenges to the Internet protocols. Remarkably, many LPWAN scenarios exhibit a multimodal round-trip time (RTT) distribution, which deviates from the common Internet RTT characteristics. This leads to a significant mismatch between retransmission timeout (RTO), computed by the standard transmission control protocol (TCP) or alternative experimental RTO algorithms, and RTT. In this article, we present the Multimodal RTO algorithm, which is able to self-adapt to the current RTT mode and produce suitable RTO values. The evaluation results show that the Multimodal RTO reduces the RTO versus RTT misalignment of the TCP RTO algorithm by an average factor of up to 5, and reduces latency in the presence of losses by up to 2 orders of magnitude, while operating safely. The Multimodal RTO is currently being considered by the IETF as a candidate mechanism for standardization.

A Sustainable Solution to Support Data Security in High Bandwidth Healthcare Remote Locations by Using TCP CUBIC Mechanism

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.

A Survey of Delay-Based and Hybrid TCP Congestion Control Algorithms

Congestion control (CC) has a significant influence on the performance of transmission control protocol (TCP) connections. Over the last three decades, many researchers have extensively studied and proposed a multitude of enhancements to standard TCP CC. However, this topic still inspires both academic and industrial research communities due to the change in Internet application requirements and the evolution of Internet technologies. The standard TCP CC infers network congestion based on packet loss events which leads to long queuing delay when bottleneck buffer size is large. A promising solution to this problem is to use the delay signal (RTT or one-way delay measurements) to infer congestion earlier and react to the congestion before the queuing delay reaches a high value. In this survey paper, we describe the delay signal and the algorithms that completely or partially utilize this type of signal. Additionally, we illustrate standard CC and modern active queue management (AQM) principles and discus the interaction between AQM and the delay signal.

Dynamic Rendering of Remote Indoor Environments Using Real-Time Point Cloud Data

Modern color and depth (RGB-D) sensing systems are capable of reconstructing convincing virtual representations of real world environments. These virtual reconstructions can be used as the foundation for virtual reality (VR) and augmented reality environments due to their high-quality visualizations. However, a main limitation of modern virtual reconstruction methods is the time it takes to incorporate new data and update the virtual reconstruction. This delay prevents the reconstruction from accurately rendering dynamic objects or portions of the environment (like an engineer performing an inspection of a machinery or laboratory space). The authors propose a multisensor method to dynamically capture objects in an indoor environment. The method automatically aligns the sensors using modern image homography techniques, leverages graphics processing units (GPUs) to process the large number of independent RGB-D data points, and renders them in real time. Incorporating and aligning multiple sensors allows a larger area to be captured from multiple angles, providing a more complete virtual representation of the physical space. Performing processing on GPU's leverages the large number of processing cores available to minimize the delay between data capture and rendering. A case study using commodity RGB-D sensors, computing hardware, and standard transmission control protocol internet connections is presented to demonstrate the viability of the proposed method.

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.

A Feedback Aware Reliable Transport Protocol with Improved Window Increment Mechanism for Inter Vehicular Wireless Network

In recent years, inter vehicular networking technologies are gaining momentum due to its merger with wireless internet through infrastructure gateways. A large quantity of internet data traffic heavily relies on transmission control protocol (TCP) due to reliable connection oriented services. TCP with sluggish window growth and weakly shaped congestion control mechanism fails under multi-hop inter vehicular conditions due to disjoint links and wireless channel errors. This paper introduces inter vehicular access network (IVAN) TCP, designed to suppress the limitations of congestion control and window increment algorithm in the multi-hop inter vehicular wireless environment. The IVAN’s flexible window increment algorithm newer growth pattern achieves a faster increase in the transmission rate. Furthermore, IVAN’s congestion control algorithm reduces the transmission rate based on sender’s window utility and relay router’s explicit congestion notification. The simulation results confirm the significant rise in IVAN’s throughput performance, substantial reduction in the queue packet drop and packet latency against standard TCP variants under inter vehicular environment.

VoIP transmission mechanism based on TCP

As the widespread employment of firewalls on the Internet, user datagram protocol (UDP) based voice over Internet protocol (VoIP) system will be unable to transmit voice data. This paper proposed a novel method to transmit voice data based on transmission control protocol (TCP). The method adopts a disorder TCP transmission strategy, which allows discontinuous data packets in TCP queues read by application layer directly without waiting for the retransmission of lost data packets. A byte stream data boundary identification algorithm based on consistent overhead byte stuffing algorithm is designed to efficiently identify complete voice data packets from disordered TCP packets arrived so as to transmit the data to the audio processing module timely. Then, by implementing the prototype system and testing, we verified that the proposed algorithm can solve the high time delay, jitter and discontinuity problems in standard TCP protocol when transmitting voice data packets, which caused by its error control and retransmission mechanism. We proved that the method proposed in this paper is effective and practical.

TCP over scarce transmission opportunity in cognitive radio networks

Transmission control protocol (TCP) is the most popular transport layer protocol for applications that require reliable and ordered data delivery essentially. In this paper we consider the deployment of TCP to secondary users (SUs) in overlay cognitive radio networks (CRNs), and address its performance degradation; in CRNs, SU’s transmissions are frequently disrupted by the detection of primary user’s transmission, and which makes the SU experience consecutive retransmission-timeout and its exponential backoff. Subsequently, the TCP in SU does not proceed with the transmission even after the disruption is over or the SU hands over to other idle spectrum. To tackle this problem, we propose a cross-layer approach called TCP-Freeze-CR; lower layer protocols send the overlying TCP two different cross-layer signals, freeze on the detection of primary user’s transmission, and unfreeze after handing over to an idle spectrum. Moreover we consider a practical situation where either secondary transmitter (ST) or secondary receiver (SR) detects primary user’s transmission; therefore additional message exchanges are needed between ST and SR to retrieve and resynchronize to other idle spectrum, i.e., spectrum synchronization. This situation is more complex than the case where both ST and SR detect primary user’s transmission. Hereby, we develop a spectrum synchronization procedure coupled with TCP-Freeze-CR into a finite state machine. All of our proposals are implemented and evaluated on a real CRN consisting of 6 software radio platforms. In the implementation, we deploy 802.15.4 implementation as a target physical layer protocol, and couple it with TCP-Freeze-CR using Unix Domain Socket. The experimental results illustrate that standard TCP suffers from significant performance degradation in CRNs, and show that TCP-Freeze-CR can greatly alleviate the degradation; e.g., for 1200 s, ST with TCP-Freeze-CR can send about 10 times more packets than ST with standard TCP.

Special issue on main results of Call‐off Order 3 (CoO3) in SatNEx III

Special issue on main results of Call‐off Order 3 (<scp>CoO3</scp>) in <scp>SatNEx</scp> III

A congestion control middleware layer with dynamic bandwidth management for satellite communications

Historically, satellites have been set aside for what regards Internet connectivity; however, the interest in their usage to provide Internet connectivity is now rising again. Because of the growing demand for Internet services around the world, satellites can be an effective medium to serve scarcely populated areas as well as mission-critical communications. While the standard transmission control protocol (TCP) performs badly when employed on satellite links for the high propagation delay, when a number of client hosts are wirelessly connected to a gateway that forwards and receives traffic across such links, the major limit is represented by the channel condition estimation performed by the TCP through loss detection and/or acknowledgement-based timing information. This paper proposes congestion control middleware layer (C2ML+), a centralized and collaborative middleware with dynamic bandwidth management, that aims to improve performance and QoS for TCP flows in the aforementioned scenarios. Results of ns-3 simulations show an improvement in aggregate throughput, a significant reduction of latencies because of low queues occupancy levels, and higher fairness and friendliness guarantees among flows. They also confirm that C2ML+ allows a dynamic and efficient usage of the bottleneck link, avoiding a waste of resources when some client nodes are unable to fully exploit their transmission potential. Copyright © 2015 John Wiley & Sons, Ltd.

An Experimental Implementation of a Cross-Layer Approach for Improving TCP Performance over Cognitive Radio Networks

In cognitive radio networks (CRNs), the performance of the transmission control protocol (TCP) at the secondary user (SU) severely drops due to the mistrigger of congestion control. A long disruption is caused by the transmission of primary user, leading to the mistrigger. In this paper, we propose a cross-layer approach, called a CR-aware scheme that enhances TCP performance at the SU. The scheme is a sender side addition to the standard TCP (i.e., TCP-NewReno), and utilizes an explicit cross-layer signal delivered from a physical (or link) layer and the signal gives an indication of detecting the primary transmission (i.e., transmission of the primary user). We evaluated our scheme by implementing it onto a software radio platform, the Universal Software Radio Peripheral (USRP), where many parts of lower layer operations (i.e., operations in a link or physical layer) run as user processes. In our implementation, we ran our CR-aware scheme over IEEE 802.15.4. Furthermore, for the purpose of comparison, we implemented a selective ACK-based local recovery scheme that helps TCP isolate congestive loss from a random loss in a wireless section.

Transmission control protocol performance enhancement for mobile broadband interactive satellite communication system: a cross‐layer approach

SummaryMobile broadband interactive satellite communication system is of great interest in both academic and industrial communities. However, the conventional strict‐layered protocol stack architecture and the standard TCP version perform poorly over satellite link. In this paper, we propose a comprehensive cross‐layer Transmission Control Protocol (TCP) optimization architecture while considering the main factors that affect the TCP performance. In our proposed architecture, we adopt two TCP split connection performance enhancing proxies to isolate the satellite link from the terrestrial part of the broadband satellite communication system. Then, based on the proposed cross‐layer architecture, we present an analytical model for the TCP throughput by taking the modulation and coding (ModCod) mode and the allocated bandwidth into account. In addition, we put forward a TCP‐driven bandwidth sharing and ModCod mode optimization algorithm to maximize the TCP throughput in satellite link. Extensive simulation results illustrate that our proposed comprehensive cross‐layer TCP optimization approach is able to improve the TCP throughput significantly. Copyright © 2014 John Wiley &amp; Sons, Ltd.

A machine learning framework for TCP round-trip time estimation

Abstract In this paper, we explore a novel approach to end-to-end round-trip time (RTT) estimation using a machine-learning technique known as the experts framework. In our proposal, each of several ‘experts’ guesses a fixed value. The weighted average of these guesses estimates the RTT, with the weights updated after every RTT measurement based on the difference between the estimated and actual RTT. Through extensive simulations, we show that the proposed machine-learning algorithm adapts very quickly to changes in the RTT. Our results show a considerable reduction in the number of retransmitted packets and an increase in goodput, especially in more heavily congested scenarios. We corroborate our results through ‘live’ experiments using an implementation of the proposed algorithm in the Linux kernel. These experiments confirm the higher RTT estimation accuracy of the machine learning approach which yields over 40% improvement when compared against both standard transmission control protocol (TCP) as well as the well known Eifel RTT estimator. To the best of our knowledge, our work is the first attempt to use on-line learning algorithms to predict network performance and, given the promising results reported here, creates the opportunity of applying on-line learning to estimate other important network variables.

Congestion Control Technique for Wireless Networks

Congestion control in wireless networks has been extensively investigated over the years and several schemes and techniques have been developed, all with the aim of improving performance in wireless net-work. With the rapid expansion and implementation of wireless technology it is essential that the congestion control problem be solved. This paper presents a congestion control schemes which are different in slow start threshold calculation, bandwidth estimation, and congestion window manipulation. A comprehensive comparison of these approaches is given in relation to assumptions, bandwidth estimation, congestion window size manipulation, performance evaluation, fairness and friendliness and improved throughput. I. Introduction The congestion control in wireless networks has been extensively investigated over the years and several schemes and techniques have been developed, all with the aim of improving performance in wireless network. With the rapid expansion and implementation of wireless technology it is essential that the congestion control problem be solved. The Transmission Control Protocol (TCP) has been widely used in today's Internet. The protocol supports reliable data transport by establishing a connection between the transmitting and receiving ends. The transmitter starts a timeout mechanism when sending a packet to the receiver. The transmitter constantly tracks the round-trip times (RTTs) for its packets as a means to determine the appropriate timeout period. At the receiver, each received packet is acknowledged implicitly or explicitly to the transmitter. If the transmitter does not receive an acknowledgment for a given packet when the corresponding timeout period expires, the packet is deemed to be lost and subject to retransmission. A congestion window with dynamically adjusted size is used by the protocol to regulate the traffic flow from the transmitter to the receiver. Although TCP was initially designed and optimized for wired networks, the growing popularity of wireless data applications has lead third generation wireless networks such as CDMA2000 and UMTS networks to extend TCP to wireless communications as well. The initial objective of TCP was to efficiently use the available bandwidth in the network and to avoid overloading the network (and the resulting packet losses) by appropriately throttling the senders' transmission rates. Network congestion is deemed to be the underlying reason for packet losses. Consequently, TCP performance is often unsatisfactory when used in wireless networks and requires various improvement techniques. A key factor causing the unsatisfactory performance is that the radio link quality in wireless networks can fluctuate greatly in time due to channel fading and user mobility 1.1 Problems of TCP in Wireless Scenario TCP protocol may experience performance degradations over wireless networks, due to non-congestion related packet loss and varying round trip times Unmodified standard TCP performs poorly in a wireless environment due to its inability to distinguish packet losses caused by network congestion from those attributed to transmission errors. Packet loss or reception of out of- order packets indicates failures. To remove such failures ,TCP implements flow control and congestion control algorithms based on the sliding window. The performance of TCP is generally lower in wireless networks than in fixed .Thesis explained by the fact that TCP cannot distinguish problems that typically occur in wireless networks from congestion. The congestion control algorithms in TCP are based on the assumptions that data is lost mainly due to congestion and that data loss duet transmission errors is rare .Therefore, data loss is interpreted as a signal of congestion in the network. Even in a wireless network, where data loss may not be related to congestion, TCP considers it as a congestion case.TCP segments may be lost if the radio conditions are poor and the link layer protocol provides a low reliability. After some retransmission attempts the link layer protocol gives up and leaves further error recovery to TCP. Handover events may also lead to data loss. A whole window of data may be lost due to handover. Data loss due to an unreliable link layer or a handover may cause a timeout event followed by slow start or three Daypack followed by fast retransmit and fast recovery. In either case, the congestion control action taken by TCP is unnecessary. Directly after the loss event ,the radio quality may become high again, and after handover data may be transmitted without problems to the new base station.TCP may also misinterpret a sudden increase in the round trip time as data loss. If the delay is long enough for the retransmission timer to expire before an acknowledgments received, then TCP misinterprets the delay as an indication of data loss due to congestion.

Packet Recycling and Delayed ACK for Improving the Performance of TCP over MANETs

Most of the schemes that were proposed to improve the performance of transmission control protocol (TCP) over mobile ad hoc networks (MANETs) are based on a feedback from the network, which can be expensive (require extra bandwidth) and unreliable. Moreover, most of these schemes consider only one cause of packet loss. They also resume operation based on the same stand-by parameters that might vary in the new route. Therefore, we propose two techniques for improving the performance of TCP over MANETs. The first one, called TCP with packet recycling (TCP-PR), allows the nodes to recycle the packets instead of dropping them after reaching the retransmission limit at the MAC layer. In the second technique, which is called TCP with adaptive delay window (TCP-ADW), the receiver delays sending TCP ACK for a certain time that is dynamically changed according to the congestion window and the trip time of the received packet. TCP-PR and TCP-ADW are simple, easy to implement, do not require network feedback, compatible with the standard TCP, and do not require distinguishing between the causes of packet loss. Our thorough simulations show that the integration of our two techniques improves the performance of TCP over MANETs.

Deploying Internet Protocol Security in satellite networks using Transmission Control Protocol Performance Enhancing Proxies

SUMMARYApplications that use the reliable Transmission Control Protocol (TCP) have a significant degradation over satellite links. This degradation is mainly a consequence of the congestion control algorithm used by standard TCP, which is not suitable for overcoming the impairments of satellite networks. To alleviate this problem, two TCP Performance Enhancing Proxies (PEPs) can be deployed at the edges of the satellite segment. Then these PEPs can use different mechanisms such as snooping, spoofing and splitting to achieve a better TCP performance. In general, these mechanisms require the manipulation of the Internet Protocol (IP) and TCP headers that generates a problem when deploying the standard IP security (IPsec) protocol. The security services that IPsec offers (encryption and/or authentication) are based on the cryptographic protection of IP datagrams, including the corresponding IP and TCP headers. As a consequence, these cryptographic protections of IPsec conflict with the mechanisms that PEPs use to enhance the TCP performance in the satellite link. In this article, we detail the reasons that cause this conflict, and we propose three different approaches to deploy IPsec in a scenario with TCP PEPs. Our proposals provide different trade‐offs between security and TCP performance in some typical scenarios that use satellite networks. Copyright © 2012 John Wiley &amp; Sons, Ltd.