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Broadband mobile networks have rapidly evolved over the last years. The unique environment they operate, however, sets new challenges for the Transmission Control Protocol (TCP). In this paper, we study the impact of Long Term Evolution (LTE) handover on TCP performance with different TCP congestion control algorithms and in a wide range of networks settings, including different handover types, random channel errors and support of error correction at lower layers. Our results suggest that the congestion control algorithms employed by TCP variants during the LTE X2 handover have different impacts on TCP performance. In particular CUBIC, Scalable TCP and BIC-TCP achieve high goodput and fast window convergence in available channel capacity after LTE X2 handover. In environments involving packet drops that are not corrected at lower layers and single errors with high probability, all TCP variants exhibit very low performance levels. Finally, in terms of fairness, our results identify fairness issues among TCP flows with different Bandwidth Delay Products (BDP), with CUBIC offering the best results.

High-speed networks are a key technology which has brought very innovative solutions to the world of computer networks while retaining mobility, scalability, reliability… It provides a very efficient service in terms of signal speed and optimality. However, such networks with such performance needs robust systems capable of adapting to the different changes that a network connection can experience, beyond this was born the need to design congestion control algorithms capable of preserving network performance while providing high speed with low latency and good signal quality. By the way, analyzing these algorithms is the main purpose of this paper.

This memo makes the case for interdependent TCP control blocks, where part of the TCP state is shared among similar concurrent connections, or across similar connection instances. TCP state includes a combination of parameters, such as connection state, current round- trip time estimates, congestion control information, and process information. This state is currently maintained on a per-connection basis in the TCP control block, but should be shared across connections to the same host. The goal is to improve transient transport performance, while maintaining backward-compatibility with existing implementations.

We study TCP flows coexistence between Massive Multiplayer Online Role Playing Games (MMORPGs) and other TCP applications, by taking World of Warcraft (WoW) and a file transfer application based on File Transfer Protocol (FTP) as an example. Our focus is on the effects of the sender buffer size and FTP cross-traffic on the queuing delay experienced by the (MMORPG) game traffic. A network scenario corresponding to a real life situation in an ADSL access network has been simulated by using NS2. Three TCP variants, namely TCP SACK, TCP New Reno, and TCP Vegas, have been considered for cross-traffic. The results show that TCP Vegas is able to maintain a constant rate while competing with the game traffic, since it prevents packet loss and high queuing delays by not increasing the sender window size. TCP SACK and TCP New Reno, on the other hand, tend to continuously increase the sender window size, thus potentially allowing higher packet loss and causing undesired delays for the game traffic. In terms of buffer size, we have established that smaller buffers are better for MMORPG applications, while larger buffers contribute to a higher overall delay.

Congestion control protocols for background data are commonly conceived and designed to emulate low priority traffic, which yields to transmission control protocol (TCP) flows. In the presence of even a few very long TCP flows, this behavior can cause bandwidth starvation, and hence, the accumulation of large numbers of background data flows for prolonged periods of time, which may ultimately have an adverse effect on the download delays of delay-sensitive TCP flows. In this paper, we look at the fundamental problem of designing congestion control protocols for background traffic with the minimum impact on short TCP flows while achieving a certain desired average throughput over time . The corresponding optimal policy under various assumptions on the available information is obtained analytically. We give tight bounds of the distance between TCP-based background transfer protocols and the optimal policy, and identify the range of system parameters for which more sophisticated congestion control makes a noticeable difference. Based on these results, we propose an access control algorithm for systems where control on aggregates of background flows can be exercised, as in file servers. Simulations of simple network topologies suggest that this type of access control performs better than protocols emulating low priority over a wide range of parameters.

Transmission Control Protocol (TCP) with Network Coding (TCP/NC) was designed to recover the lost packets without TCP retransmission to improve the goodput performance in lossy networks. However, TCP/NC is too costly to be implemented in some types of end devices, e.g., with less memory and power. In addition, TCP/NC across loss-free but thin networks may waste scarce link bandwidth due to the redundant combination packets sacrificed for the lossy network. In this paper, we propose the TCP/NC tunnel to convey end-to-end TCP sessions on a single TCP/NC flow traversing a lossy network between two special gateways without per-flow management. We implemented and validated our proposal in Network Simulator 3, in which each gateway runs a reinforced version of TCP/NC that we previously developed. The results show that the proposed TCP/NC tunnel can mitigate the goodput degradation of end-to-end TCP sessions traversing a lossy network without any change in TCP on each end host., The 22nd IEEE Symposium on Computers and Communications (ISCC'17), 03 - 06 July 2017, Heraklion, Crete, Greece

Smart grid communication networks adopt a variety of communication technologies interconnecting numerous and diverse equipment. The requirement of supporting a large traffic volume over such networks efficiently, reliably, and fairly among various applications calls for the study of the effectiveness of the transmission control protocol (TCP). The question raised in this paper is whether various TCP variants perform differently. We are particularly interested in comparing the performance of TCP-Reno and TCP-Vegas variants when handling sensory and metering traffic. The former variant follows a loss-based congestion control mechanism while the latter adds delay-based enhancements. Contrary to expectations, our simulation and analytical comparison results show that TCP-Reno outperforms TCP-Vegas in terms of packet loss rate. However, to take advantage of Vegas’ added features, a previously proposed scheme named split- and aggregated-TCP (SA-TCP) is recommended. The scheme splits meters’ TCP connections at intermediate devices and forwards collected data over an aggregated TCP connection to a data center.

The global mobile traffic is growing exponentially, and currently more than 90 percent of the internet traffic depends on TCP for reliable transmission. However, it is well known that TCP performs quite poorly over unreliable wireless networks, while its dynamic TCP flow control is sensitive to congestion events and tends to underutilize the available network capacity. Despite these known limitations, the industry believes that significant network performance enhancement can be achieved through TCP optimization, and in particular through solutions centered on Split-TCP. In this article we share a sample of TCP throughput performance measurements after implementing Split-TCP on commercial 4G LTE and 3G networks. Experimental results show that Split-TCP provides average TCP throughput gains on the order of 60 percent over a 4G LTE network and on the order of 5 percent over a 3G network. Furthermore, TCP throughput gains superior to 200 percent were measured for individual TCP connections over the 4G LTE network. We expect such throughput gains to be even greater when advanced radio technologies, such as 5G, are deployed. For these reasons, Split-TCP has the potential to be widely deployed in both current and next generation networks.

Background/Objectives: The working of TCP upon Wireless Networks is the most difficult issue, as a result of arbitrary misfortunes and ACK impedance. TCP likewise experiences execution declination as far as making postponement and overhead in system due to the poor attributes of the Wireless channel. Keeping in mind the end goal is to beat these issues and to enhance the performance of TCP in MANETs. Methods/Statistical Analysis: The Congestion Control in Multi - Hop Mobile Ad - Hoc Networks Using Cross - Layer Based TCP Protocol Approach, the mechanism triggers congestion whenever the Channel Occupied Ratio (COR) reaches a maximum threshold value and the received signal strength is less than a minimum threshold value. Following it, congestion control scheme controls the data sending rate of the sender by determining the available bandwidth, delay of its link and COR. Further, a fair resource allocation scheme is put forwarded. Findings: A Smart Acknowledgment Distributed Channel Access (SADCA) is proposed, where a different Access Category (AC) for data less TCP acknowledgement packets is utilized and after that it is relegated with the highest priority. Along these lines, the postponement amid the transmission of packet can be lessened and furthermore the affirmation for the packet can be sent instantly. Keeping in mind the end goal to increase the performance of TCP, delay window size can likewise be balanced by considering the parameters, for example, transmission rate, number of jumps, and Channel Occupied Ratio (COR). Thus, the proposed plan helps in staying away from any sort of postponement and overhead to send the TCP acknowledgements. Applications/Improvements: The delay during the transmission of packet is decreased and the packet is recognized instantly. To expand the performance, the delay window size can be balanced for improvement purpose by considering the parameters, for example, transmission rate, number of jumps, and COR. The proposed scheme improves delay and overhead to send TCP acknowledgement.

Closed-loop flow control protocols, such as the prominent implementation transmission control protocol (TCP), are prevalent in the Internet, today. TCP has continuously been improved for greedy traffic sources to achieve high throughput over networks with large bandwidth delay products. Recently, the increasing use for streaming and interactive applications, such as voice and video, has shifted the focus toward its delay performance. Given the need for real-time communication of non-greedy sources via TCP, we present an estimation method for performance evaluation of closed-loop flow control protocols. We characterize an end-to-end connection by a service curve that provides statistical guarantees for arbitrary traffic. The estimation is based on end-to-end measurements at the application level that include all effects induced by the network and by the protocol stacks of the end systems. From our measurements, we identify different causes for delays. We show that significant delays are due to queueing in protocol stacks. Notably, this occurs even if the utilization is moderate. Using our estimation method, we compare the impact of fundamental mechanisms of TCP on delays at the application level: in detail, we analyze parameters relevant for network dimensioning, including buffer provisioning and active queue management, and parameters for server configuration, such as the congestion control algorithm. By applying our method as a benchmark, we find that a good selection can largely improve the delay performance of TCP.

TCP remains the dominant transport protocol for Internet traffic. It is usually considered to have its sending rate covered by a sliding window congestion control mechanism. However, in addition to this normal congestion control, a number of other mechanisms limit TCP throughput. This paper analyses the extent to which network, host and application settings define flow throughput over time and across autonomous systems. Our study draws on data from a longitudinal study spanning five years of passive traces collected from a single transit link. Mechanisms for this include limiting by application, interference with the TCP window control mechanism and artificial limitations on maximum window sizes by the operating system. This paper uses a large data set to assess the impact of each mechanism. We conclude that more than half of all heavy-hitter inbound traffic remains throttled by constraints beyond network capacity. For this data set, TCP congestion control is no longer the dominant mechanism that moderates throughput.

Current and future communication scenarios are very different from those in which TCP was conceived and developed, bringing new protocol requirements. Performance optimization is usually pursued with patching of the traditional TCP implementations. Following an alternative approach, TCP Wave has been designed with the primary goal to satisfy new requirements coming from current networks, such as adaptation to bandwidth/delay changes (due to mobility, dynamic switching, handover), efficient management of spurious losses as well as link interruption, optimal exploitation of very high link capacity and efficient transmission of small objects, as in most of Web applications (Web browsing, sensor apps, SCADA, M2M, etc.), irrespective of the underlying link characteristics. Protocol fairness, TCP friendliness, congestion and flow control and error recovery are also guaranteed. TCP Wave replaces the window-based transmission paradigm of the standard TCP with a burst-based transmission, the ACK-clock scheduling with a self-managed timer and the RTT-based congestion control loop with an Ack-based Capacity and Congestion Estimation (ACCE) module. As a valuable study case, the novel TCP Wave capabilities has been validated on a generic NS-3 simulation testbed where all the most challenging events impacting on transport protocol can occur.

Cloud gaming has emerged as a promising application to enable high-end game playing with thin clients. Transmission control protocol (TCP) is pervasively adopted as the transport-layer protocol in the mainstream cloud gaming systems for video communication. However, streaming mobile cloud gaming video using the TCP is challenged with several key technical barriers: 1) the performance limitations of wireless networks in bandwidth and reliability; 2) the high throughput demand and stringent delay constraint imposed by high-quality gaming video transmission; and 3) the deadline violations and throughput fluctuations caused by the packet retransmission and congestion control mechanisms in the TCP. To address these critical problems, this paper proposes an application-layer source–forward error correction (FEC) coding framework dubbed adaptive source–FEC coding over TCP (ESCOT). First, we analytically formulate the optimization problem of joint source–FEC coding to minimize the end-to-end distortion of real-time video communication over TCP. Second, we develop a heuristic solution for effective loss rate approximation, source rate control, and FEC coding adaptation. ESCOT is distinct from existing source–FEC coding schemes in proactively analyzing and leveraging the TCP characteristics. The proposed solution is able to effectively mitigate both consecutive and sporadic video frame drops caused by congestion and random packet losses. We conduct the performance evaluation through extensive emulations in the Exata platform using real-time gaming video encoded by the H.264 codec. Experimental results show that the ESCOT advances the state of the art with noticeable improvements in video peak signal-to-noise ratio, end-to-end delay, goodput, and frame success rate.

Existing congestion control mechanisms cannot feedback network capacity accurately to traffic sources. In this paper, we propose Adjusting Advertised Window (AAW), an accurate congestion control mechanism, based on programmable networks. After theoretical analysis of the performance of AAW, we explain the design principles. We implement AAW in P4 at intermediate nodes by modifying the advertisedWindow field at TCP headers to feedback available network capacity to traffic sources. Experiments show that AAW gets 5% and 10% increase in throughput compared to HSTCP and Reno respectively. AAW can also solve ECN-non-ECN unfairness and compared with vCC, it gains 16% increase in throughput.

As the exponential growth of the Internet, there is an increasing need to provide different types of services for numerous applications. Among these services, low-priority data transfer across wide area network has attracted much attention and has been used in a number of applications, such as data backup and system updating. Although the design of low-priority data transfer has been investigated adequately in low speed networks at transport layer, it becomes more challenging for the design of low-priority data transfer with the adaptation to high bandwidth delay product networks than the previous ones. This paper proposes an adaptive low-priority protocol to achieve high utilization and fair sharing of links in high bandwidth delay product networks, which is implemented at transport layer with an end-to-end approach. The designed protocol implements an adaptive congestion control mechanism to adjust the congestion window size by appropriate amount of spare bandwidth. The improved congestion mechanism is intent to make as much use of the available bandwidth without disturbing the regular transfer as possible. Experiments demonstrate that the adaptive low-priority protocol achieve efficient and fair bandwidth utilization, and remain non-intrusive to high priority traffic. Copyright © 2016 John Wiley & Sons, Ltd.

Wireless networks and large bandwidth delay product (BDP) networks are two types of challenging environments for TCP congestion control. Many congestion control algorithms have been proposed to improve the performance of TCPs in these two environments. Although these improved algorithms can achieve remarkable performance enhancements for either of the two environments, designing a congestion algorithm that performs well over heterogeneous networks that contain both wireless and large BDP links remains a great challenge. In this study, we propose a novel congestion avoidance algorithm called TCP-FIT, which can perform excellently over both wireless and large BDP links while maintaining good fairness with the standard TCP Reno algorithm. To evaluate the proposed algorithm, theoretical analysis is presented for equilibrium, network utilization, stability, TCP friendliness, RTT fairness, and responsiveness. A series of experimental results obtained using network emulators and over the "live" Internet are also presented to demonstrate the significant performance and fairness improvements of TCP-FIT compared with other state-of-the-art algorithms.

Multipath TCP is a major extension of TCP, designed for leveraging the increasing availability of multiple interfaces in end hosts, on one side, and the existence of diverse Internet paths between hosts, on the other. This paper proposes a measurement methodology and provides a first evaluation, based on real Internet experiments, of the user benefit of using MPTCP instead of TCP in devices with multiple wireless/wired networking interfaces. We focus on bandwidth utilization and file transfer delays. Our experiments, on a testbed with two disjoint paths connecting a server and a dual-homed probe, indicate that MPTCP is able, in most cases, to take advantage of additional bandwidth with limited cost in terms of delay, but also show that the MPTCP bandwidth benefit substantially degrades when the interfaces have very different bandwidth capacities.

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.

Unmanned aerial vehicles (UAVs) have recently become popular for both recreational and commercial use and UAV networks have thus started to attract the attention of researchers in area of the computer communication and networking. One important topic in UAV networks is congestion control because congestion causes packet losses and delays which result in the waste of all types of network resources such as bandwidth and power. Although there are studies on the performance of TCP congestion control in wireless networks, they focus on terrestrial networks of two dimensions in general. In this paper we study the performance of TCP congestion control in three dimensional UAV networks. In particular, we investigate how TCP congestion control performs in such type of network using various radio propagation models. Our data on the average flow throughput, packet delay, and packet loss rate in UAV networks show that TCP congestion control improves the network performance of UAV networks in general, but it faces challenges when the link losses become severe. Our study thus shows that investigation on new congestion control schemes is stilled needed for the emerging UAV networks.

While today's networks have been shown to exhibit HBDP (High Bandwidth Delay Product) characteristics, the legacy TCP increases the size of the congestion window slowly and decreases the size of the congestion window drastically such that it is not suitable for HBDP Networks. In order to solve this problem with the legacy TCP, many congestion control TCP mechanisms have been proposed. C-TCP (Compound-TCP) is a hybrid TCP which is a synergy of delay-based and loss-based approaches. C-TCP adapts the decreasing rate of the delay window without considering the congestion level, leading to degradation of performance. In this paper, we propose a new scheme to improve the performance of C-TCP. By controlling the increasing and decreasing rates according to the congestion level of the network, our proposed scheme can improve the bandwidth occupancy and fairness of C-TCP. Through performance evaluation, we show that our proposed scheme offers better performance in HBDP networks as compared to the legacy C-TCP.

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.

Cloud computing provides various diverse services for users accessing big data through high data rate cellular networks, e.g., LTE-A, IEEE 802.11ac, etc. Although LTE-A supports very high data rate, multi-hop relaying, and cooperative transmission, LTE-A suffers from high interference, path loss, high mobility, etc. Additionally, the accesses of cloud computing services need the transport layer protocols (e.g., TCP, UDP, and streaming) for achieving end-to-end transmissions. Clearly, the transmission QoS is significantly degraded when the big data transmissions are done through the TCP protocol over a high interference LTE-A environment. The issue of providing high data rate and high reliability transmissions in cloud computing needs to be addressed completely. Thus, this paper proposes a cross-layer-based adaptive TCP algorithm to gather the LTE-A network states (e.g., AMC, CQI, relay link state, available bandwidth, etc.), and then feeds the state information back to the TCP sender for accurately executing the network congestion control of TCP. As a result, by using the accurate TCP congestion window (cwnd) under a high interference LTE-A, the number of timeouts and packet losses are significantly decreased. Numerical results demonstrate that the proposed approach outperforms the compared approaches in goodput and fairness, especially in high interference environment. Especially, the goodput of the proposed approach is 139.42 % higher than that of NewReno when the wireless loss increases up to 4 %. Furthermore, the throughput and the response functions are mathematically analyzed. The analysis results can justify the claims of the proposed approach.

These days, the networks have exhibited HBDP (High Bandwidth Delay Product) characteristics. The legacy TCP slowly increases the size of the congestion window and drastically decreases the size of a congestion window. The legacy TCP has been found to be unsuitable for HBDP networks. TCP mechanisms for solving the problems of the legacy TCP can be categorized into the loss-based TCP and the delay-based TCP. Most of the TCP mechanisms use the standard slow start phase, which leads to a heavy packet loss event caused by the overshoot. Also, in the case of congestion avoidance, the loss-based TCP has shown problems of wastage in terms of the bandwidth and RTT (Round Trip Time) fairness. The delay-based TCP has shown a slow increase in speed and low occupancy of the bandwidth. In this paper, we propose a new scheme for improving the over shoot, increasing the speed of the bandwidth and overcoming the bandwidth occupancy and RTT fairness issues. By monitoring the buffer condition in the bottleneck link, the proposed scheme does congestion control and solves problems of slow start and congestion avoidance. By evaluating performance, we prove that our proposed scheme offers better performance in HBDP networks compared to the previous TCP mechanisms.

We identified a problem that the current TCP variants suffer from synchronization caused by tiny buffers at high speed networks.We proposed and implement a new TCP that explores parallelism among multiple channels and TCP desynchronization for high speed networks.We evaluated the performance of the new DMCTCP over different networking environment.We analyzed the optimal performance of the proposed DMCTCP based on theoretical analysis. The buffer sizing problem is a big challenge for high speed network routers to reduce buffer cost without throughput loss. The past few years have witnessed debate on how to improve link utilization of high speed networks where the router buffer size is idealized into dozens of packets. Theoretically, the buffer size can be shrunk by more than 100 times. Under this scenario, widely argued proposals for TCP traffic to achieve acceptable link capacities mandate three necessary conditions: over-provisioned core link bandwidth, non-bursty flows, and tens of thousands of asynchronous flows. However, in high speed networks where these conditions are insufficient, TCP traffic suffers severely from routers with tiny buffers.To explore better performance, we propose a new congestion control algorithm called Desynchronized Multi-Channel TCP (DMCTCP) that creates a flow with parallel channels. These channels can desynchronize each other to avoid TCP loss synchronization, and they can avoid traffic penalties from burst losses. Over a 10?Gb/s large delay network ruled by tiny buffer routers, our emulation results show that bottleneck link utilization can reach over 80% with much fewer number of flows. Compared with other TCP congestion control variants, DMCTCP can also achieve much better performance in high loss rate networks. Facing the buffer sizing challenge, our study is a new step towards the deployment of optical packet switching networks.

the era of communication congestion is the majorproblemwhich degrades the performance. In Ad hoc networks due to infrastructure less character there is no guaranteed bandwidth and node mobility. The traditional TCP variants were good for wired environment, whereas in Ad-hoc networks there are lots of other factors to be considered. The congestion window is one of the factors that determine the number of bytes that can be outstanding at any time. The congestion window indicates the maximum amount of data that can be sent out on a connection without being acknowledged. HSTCP is the very important protocol that will be used only when congestion window increases abruptly. To improve the retransmission time and to avoid congestion in completely dynamic conditions where node mobility is completely random is the main concern of our research work. KeywordsHSTCP, Congestion, MANETs

The Transmission Control Protocol (TCP) has been profusely used by diverse applications such as FTP, email, and HTTP. In recent decades, numerous TCP variants have been developed to fit the fast increasing of network capacities and improve the performance of TCP algorithms in various scenarios, particularly in the high-bandwidth-delay-product (high-BDP) and lossy networks. Although the performance improvements are different for different TCP under different networking conditions, implementing a new congestion control algorithm that is suitable for a wide range of network conditions is still a challenge. In this letter, we propose a novel scheme, i.e., the congestion-level-based TCP (CLTCP), which could perform effectively in both high-BDP and lossy networks. Different from the TCP-FIT, the CLTCP does not use the delay variation but the congestion level as a signal to control the number of virtual parallel flows in a TCP connection. This could avoid the influence of the delay measurement error in the TCP-FIT. Extensive experiments in NS-2 show that the performance of the CLTCP is significantly improved, as compared to other state-of-the-art algorithms, while maintaining good fairness.

Summary 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.

The final publication is available at Springer via http://dx.doi.org/10.1007/s11227-015-1478-9, Traffic analysis is an essential part of capacity planning, quality of service assurance, and reinforcement of security in current telecommunication networks. Traffic volume increases with network speed and the analysis of large traffic traces is computationally intensive. The paper presents, for the first time ever, a flow extraction software that allows to obtain complex TCP-aware flow records at 4.4 millions of packets per second in a single GPU. Such TCP flow records include number of retransmissions and duplicates per flow, whose calculation is very challenging to obtain at high-speed. Our software significantly increases the processing performance of the recently proposed high-speed sniffers based on commodity hardware and demonstrates the advantages of applying massively parallel processing devices for traffic analysis.

Transmission Control Protocol (TCP) in infrastructure based vehicular networks is dedicated to support reliable Internet services for mobile users. However, an end-toend TCP flow not only experiences some common challenges in wireless mobile networks, such as high packet loss rate, medium access competition, unstable wireless bandwidth, and dynamic topology, etc., but also suffers from performance degradation due to traffic congestion at the Road-Side Units (RSUs) that connect the wireline and wireless networks. In order to address the challenging issues related to reliable TCP transmissions in infrastructure based vehicular networks, we propose an RSU based TCP (R-TCP) scheme. For wireline source nodes, R-TCP adopts a novel flow control mechanism to adjust transmission rates according to the status of bottleneck link. Specifically, during the short wireless connection time in Infrastructure based vehicular networks, R-TCP quickly chooses an ideal transmission rate for data transmissions instead of activating the slow start algorithm after the connection is established, and successfully avoids the oscillation of the transmission rate. Simulation results show that R-TCP achieves great advantages than some relate proposals in terms of throughput, end-to-end delay, and packet loss rate.

With the development of Internet, various kinds of new applications appear constantly. They all have high requirements to the time delay, throughput, especially strong real-time applications such as mobile monitoring, video calls. The satellite network in Navigation Satellite System, which is necessary for the mobile monitoring, has many disadvantages such as asymmetric bandwidth, unstable network, high bit error rate and so on. This is a new challenge to the existing congestion control method. In order to improve the performance of congestion control mechanism, we put forward TCP-QS (Quick Start) from the TCP protocol control in this paper. TCP-QS algorithm mainly optimize the slow start stage. At the beginning of the connection, the value of parameter cwnd is set as a larger value according to the detected network bandwidth in which way, the time of the slow start stage is shortened during the transmission, and is adjusted the value of parameter ssthresh dynamically according to the change of network. When packet loss occurs, it takes different methods according to the different reasons.

Cognitive radio (CR) has emerged as a promising solution to enhance spectrum utilization. In CR networks (CRNs), the secondary users can opportunistically exploit frequency bands when the primary users (PUs) do not occupy the bands. In this communication paradigm, transmission control protocol (TCP) performance may suffer from significant degradation due to the features of CRNs. In this paper, we investigate the limitations of TCP in multi-channel multi-radio multi-hop CRNs, and propose a novel TCP called TCP Network Coding Dynamic Generation Size Adjustment (TCPNC-DGSA) based on network coding. We dynamically adjust generation size in network coding operation according to the wireless communication environment--Generation Round Trip Time. In the meanwhile, we modify the TCP mechanism to fit into CRNs, by considering spectrum sensing state, spectrum changing state and presence of PUs. The simulation results indicate that TCPNC-DGSA can significantly improve the network performance in terms of throughput, bandwidth efficiency and delay. To the best of our knowledge, TCPNC-DGSA is the first TCP for CRNs from a network coding perspective, which can guarantee a quality of service in terms of delay.

TCP congestion control protocols have low performances in high speed wide area networks mainly because their slow response with large congestion windows. This TCP behavior has initiated new design phase of alternative protocols that provide improved traffic utilization in high bandwidth delay product networks. The paper presents survey of various high speed sender side congestion control proposals that preserve the fundamental host to host principle. Solutions focus on variety of problems occurring in high speed environment with intention to eliminate congestion collapses and to ensure effective resource utilization. Internet data transfer does not depend only on that how TCP will utilize the network capacity, we have to stress that TCP must cooperate with existing transmitting data protocols through the same network in order to assure fair resource sharing. Part of the paper scope are state of the art high speed TCP proposals, we explore their congestion control techniques, strengths, weaknesses and we try to detect future TCP development possibilities.

The article describes TCP TIPS, the transport level protocol, which aim is to effi- ciently use the available bandwidth. This protocol implements the proactive methods for congestion avoidance. These methods allow TCP TIPS to yield the required amount of bandwidth to high-priority flows with realtime data and to use this amount, when it becomes available.

With the proliferation of new wireless service, scarce wireless resources is expected to become a critical issue. For this reason, cognitive radio mobile ad hoc networks (CogMANET) are being developed as a promising solution to this problem. However, in CogMANET, channel switching is inherently necessary whenever a primary user with a license appears on the channel. Allowing secondary users to choose an available channel from among a wide spectrum range thus enables reliable communication in this context, but communication characteristics such as bottleneck bandwidth and RTT will change with channel switch. In response to this change, TCP has to adaptively update its congestion window ( cwnd ) to make an efficient use of the available resources. For this purpose, TCP CRAHN was proposed for CogMANET. In this paper, TCP CRAHN is first evaluated in cases where bottleneck bandwidth and RTT drastically change. Based on these results, TCP CoBA is proposed to further improve the throughput of the above use cases. TCP CoBA updates the cwnd based upon the available buffer space in the relay node upon channel switch, as well as other communication characteristics. Through simulations, we show that compared with TCP CRAHN, TCP CoBA improves the throughput by up to 200 percent.

control protocol (TCP) was initially designed for static networks to provide the consistent data delivery. The enhancement of TCP performance was also achieved with different types of networks with the introduction of new TCP deviations. However, there are still many factors that affect the performance of TCP. Mobility is one of the primary performance-affecting drivers in heterogeneous network. The research of state in this paper is to determine the best TCP variant from the mobility point of view. We simulate some TCP variants such as TCP-Tahoe, TCP-Reno, TCP-New Reno, TCP- Vegas, TCP-SACK and TCP-Westwood from the mobility point of view. The scenarios presented in this paper are supported by on- demand link weight (ODLW) routing protocol that helps find the efficient route with Quality of Service (QoS) provisioning. The scenarios are designed for the walking persons to vehicles, and particularly suited for rocky and deserted areas. To demonstrate the strength of these TCP variants, the scenarios are simulated and evaluated the QoS parameters such as round trip time (RTT) fairness, end-to-end delay, and the number of broken links. Finally, based on the outcomes, we identify the best TCP variant that could be used for several applications in the future over heterogeneous wireless networks.

TCP has always been the most prominent protocol used in internet. More than 90 % of the HTTP session in Internet has less than 20 KB data to send (Comput Commun Rev 40(3):26---33, 2010) and finish the whole transaction in the slow start phase of TCP. These Short TCP transactions face high latency and network underutilization problem in bottleneck last mile wireless broadband connections due to higher Bit Error Rate in wireless when compared with wired connection . In this paper, we evaluate Short TCP by simulating it over wireless broadband networks to bring out the open challenges in optimizing Short TCP's completion time. We also derive mathematical model for estimating and optimizing the completion time of Short TCP connections over broadband wireless networks. We extensively examine our model for Short TCP and propose a generic link level scheduler mechanism with O(1) complexity that uses cross layer information to optimize the completion time of connection. Our result shows that the mathematical model matches the simulation results and we also show that the proposed novel scheduler speeds up Short TCP connection up to 13 % which in turn increases the Quality of Service of the end user without affecting other connections.

The article describes TCP TIPS, the transport layer protocol, which aim is to efficiently use the available bandwidth. This protocol implements the proactive methods for congestion avoidance. These methods allow TCP TIPS to yield the required amount of bandwidth to high-priority flows with realtime data and to use this amount, when it becomes available.

TCP does not work well in modern cellular networks because the current congestion-window-based (cwnd-based) congestion control mechanism intimately couples congestion control and packet dispatch, which provides TCP with only indirect control of the effective data rate. The throughput degradation arising from the cwnd-based mechanism is especially serious when the uplink is congested. We describe PropRate, a new rate-based TCP algorithm that directly regulates the packets in the bottleneckbuffer to achieve a trade-off in terms of delay and throughput along a more efficient frontier than conventional cwnd-based TCP variants. To the best of our knowledge, PropRate is the first TCP algorithm that allows an application to set and achieve a target average latency, if the network conditions allow for it. Also, unlike the cwnd-based TCP mechanism, our new rate-based TCP mechanism is significantly more resilient to saturated uplinks in cellular networks. PropRate does not require modifications at the receiver and is amenable to practical deployment in the base stations and proxies in mobile cellular networks.

While protecting a target network for detecting a potential network attack, based on attack signature scanning methodology, stateful inspection plays a vital role to detect protocol-based scanning of sessions thus reducing false positives. We propose an architecture which comprises of session table management scheme to perform stateful packet inspection in real-time network scenarios. The architecture uses efficient data structure to store session information and a methodology to retrieve and modify the protocol state information. The state table also considers flow-based information for more accurately extracting attack signature parameters and thus enhancing accuracy of signature-based detection. A parallel thread for scanning the session table and for dealing with expired sessions is also incorporated to avoid memory overflow scenarios and supports higher number of valid sessions in real-time networks. A methodology to communicate with deep packet inspection engine and terminate those TCP sessions for which attack is detected which is incorporated using multithreading approach. We demonstrate two major architectures to enhance the performance of DPI with stateful inspection enabled in it using multi-processing techniques to achieve parallel processing and the experimental results.

Increasing use of online backup services, as well as the popularity of user-generated content, has increased the demand for bandwidth. However, traffic generated by these applications can impact on the responsiveness of delay-sensitive applications if they receive a 'fair-share' of the available bandwidth. Less-than-Best-Effort TCP congestion control mechanisms aim to allow lower-priority applications to utilise excess bandwidth with minimum impact to regular TCP traffic. We evaluated the performance of six Less-than-Best-Effort congestion control algorithms in different scenarios in a Linux testbed, only three of which had existing implementations for modern operating systems. The findings of this study suggest that Nice provides background throughput comparable to that of regular TCP, while maintaining low queuing delay, while CAIA Delay-Gradient (CDG) has the least impact on regular TCP traffic, at the expense of reduced throughput.

The innovative congestion control algorithm named FASTRA (Fast Active Stability TCP) is aimed for high-speed long-latency networks. Four major difficulties in FASTRA are highlighted at both packet and flow levels. The architecture and characterization of equilibrium and stability properties of FASTRA are robust. Experimental results of FASTRA outsmart TCP Reno, HSTCP, and STCP in terms of throughput, fairness, stability, and responsiveness. FASTRA aims to rapidly stabilize high-speed long-latency networks into steady, efficient and fair operating points, in dynamic sharing environments, and the preliminary results are produced as output of our project. The Proposed architecture is explained with the help of an existing real-time example as to explain why FASTRA download is chosen rather than FTP download. The Paper is concluded with the results of the new congestion control algorithm aided with the graphs obtained during its simulation in NS2. On proper implementation, many safe, FASTRA downloads and data transfers can be carried over a high speed internet network.

Mobile devices have become quite popular, and they are responsible for a significant part of Internet traffic. In these devices, multimedia services, such as video streaming over HTTP, are commonly adopted, and TCP protocol is the standard to assure reliable data transmission. To deal with applications that require higher bandwidth, several modifications were made on TCP, and they are called extensions for high performance. For instance, these extensions allow a better utilization of network bandwidth offered by high-speed network technologies, such as WCDMA (3G) and LTE (4G). Changes in network technologies and TCP parameters may solve performance issues, but this modification may increase energy consumption (which considerably impacts mobile devices). This paper presents a performance and energy consumption evaluation of network technology and TCP/IP parameters for mobile devices. We utilize video streaming over HTTP as the workload, and results indicate some TCP parameters may influence energy consumption as well as performance.

Upload and download traffic often coexist in mobile networks. However, TCP download throughput could be substantially degraded by upload traffic even if the downlink is not the bottleneck. Previous works such as RSFC and TCP-RRE can substantially improve TCP download throughput in the presence of concurrent TCP upload flows, albeit at the expense of significantly degraded upload throughput performance. This work addresses this limitation by developing a novel Aggregate Transmission Rate Controller with Upload and Download flows aggregations (ATRC-UD) to jointly accelerate concurrent TCP upload and download flows from/to the same mobile device. The insight is that existing TCP as well as other flow-based approaches all suffer from ACK packets delayed by data packets from TCP flows in the opposite direction, resulting in significant errors in bandwidth estimation. By contrast, ATRC-UD exploits data packets of the opposite direction to enable continuously estimation of the downlink bandwidth and queueing delay even when ACK packets are significantly delayed. This allows ATRC-UD to track the bandwidth and delay variations more closely to maintain a shorter queue length at the downlink, thus jointly improve the download-upload throughput. Extensive emulated and real-world experiments showed that ATRC-UD enables TCP to achieve 96% downlink bandwidth utilization while improving uplink bandwidth utilization by over 115% compared to existing approaches, such as TCP-RRE and RSFC.

Current Internet traffic is carried mainly by the Transmission Control Protocol (TCP), and the performance of the Internet depends strongly on how well TCP works. Detecting and mitigating the congestion is one of the main tasks of this protocol, in fact, different TCP versions are defined by their congestion control algorithm. Also, Active Queue Management (AQM) algorithms share the same goal of congestion mitigation with TCP, in particular, the most effective congestion control occurs when AQM and TCP work together. This paper presents a brief survey and a cross-comparison of the latest and most important TCP and AQM variants, then provides an evaluation of a different kind of performance (multiple Round-Trip Time, long delay, different congestion level, etc.) on the ns-3 network simulator over various types of environments. In a shared bottleneck, the choice of the TCP-AQM couple to adopt is crucial; we show that the results are not univocal, the “golden couple” depends on the performance that we want to boost and on the environment that we have to deal with.