Transmission Control Protocol Congestion Window Research Articles

Cross-layer modeling and analysis of TCP communications over turbulent FSO links

This paper addresses the accurate characterization of the performance of transmission control protocol (TCP) for end-to-end transport services over free-space optical (FSO) links. Unlike previous works on this topic, we introduce a second-order Markovian assumption for the variation of the TCP congestion window to capture the memory from turbulence channels and further propose a Markov chain model that maps two consecutive congestion windows along time into the state space to interpret TCP operations. The analytical expression of steady-state probability distribution vector of the proposed model in general cases where relay routers exist behind the TCP connection is derived, based on which the throughput efficiency of TCP is formulated. The high accuracy of the proposed model is verified by Monte-Carlo simulations and experiments. We further analyze the impact of different link/channel parameters on the TCP performance, and discuss briefly the effectiveness of bundle protocol-based schemes for improving the performance of FSO networks from a transport layer perspective.

An INT-based TCP window modulator for congestion control in data center networks

Modern data centers carry a huge number of diverse services and applications, which are developed by Transmission Control Protocol (TCP). The early TCP versions, such as Reno, were designed decades ago and not efficient enough to accommodate the data center environment, especially congestion control algorithms (CCAs). Data center TCP (DCTCP) was the classical CCA, which proposed for the data center networks (DCNs). All the conventional CCAs use packet loss and/or delay as congestion signals, however, these signals are not accuracy enough to conduct the CCAs to compute appropriate congestion window. In this paper, the In-band Network Telemetry (INT) data is used as congestion signal. The INT data can indicate the congestion location and level, which is more accuracy than end-to-end signals. Based on the advanced signal, an INT-based Receiver-driven TCP congestion window modulator, named IRTCP, is proposed and implemented as a loadable Linux kernel module, without any modification of the TCP/IP stack. Either in small-scale Mininet emulation or in large-scale NS3 simulation, the results show that IRTCP could provide significant performance improvement. The average FCT of IRTCP for mice flows is about 89.5%, 91.6%, 91.7% lower than Cubic-RED, Cubic-FIFO, and DCTCP. The average throughput of IRTCP for elephant flows is about 11.5%, 31%, 20.6% higher than DCTCP, Cubic-RED, Cubic-FIFO.

Playback experience driven cross layer optimisation of APP, transport and MAC layer for video clients over long‐term evolution system

In traditional communication system, information of APP (Application) layer, transport layer and MAC (Media Access Control)layer has not been fully interacted,which inevitably leads to inconsistencies among TCP congestion state, clients'requirements and resource allocation. To solve the problem, we propose a joint optimization framework, which consists of APP layer, transport layer and MAC layer, to improve the video clients'playback experience and system throughput. First, a client requirement aware autonomous packet drop strategy, based on packet importance, channel condition and playback status, is developed to decrease the network load and the probability of rebuffering events. Further, TCP (Transmission Control Protocol) state aware downlink and uplink resource allocation schemes are proposed to achieve smooth video transmission and steady ACK (Acknowledgement) feedback respectively. For downlink scheme, maximum transmission capacity requirement for each client is calculated based on feedback ACK information from transport layer to avoid allocating excessive resource to the client, whose ACK feedback is blocked due to bad uplink channel condition. For uplink scheme, information of RTO (Retransmission Timeout) and TCP congestion window are utilized to indicate ACK scheduling priority. The simulation results show that our algorithm can signficantly improve the system throughput and the clients'playback continuity with acceptable video quality.

Comparative analysis of cumulative distribution function for TCP congestion window size and triple‐duplicate period

SummaryVarious version of Transmission Control Protocol (TCP) congestion window model are available in the literature. TCP congestion control is handled in two phases: (a) slow start and (b) congestion avoidance. This paper deals with congestion avoidance phase that is based on additive increase multiplicative decrease (AIMD) mechanism in which window size either increase by one or cuts to half of the previous window size. For distribution of window sizes, a model developed by Yan et al has been presented for two loss indications: (a) triple duplicate (TD) and (b) time‐outs (TOs) only while it does not deal with the window size limitation. It does not allow to cap over the TCP congestion window size if the buffer size is restricted to a fixed value though window size could not move beyond that. The unconstrained window size moves on in the presence of low loss probability. In this paper, we propose a model for the case where the window size is bounded by a maximum value that makes applicable to window size limitation as well. Further, the work has been extended to develop a new model that obtains a cumulative distribution function for TD periods (TDPs). The proposed model is validated on ns‐2, and we conclude that observed results for distribution function are very closed to our proposed model.

Low-High Burst: A Double Potency Varying-RTT Based Full-Buffer Shrew Attack Model

The full-buffer Shrew (FB-Shrew) denial of service (DoS) attack is a variant of the classic Shrew attack that exploits the congestion control mechanism of transmission control protocol (TCP). Here, an attacker sends a high-rate burst of attack packets only after the router buffer is filled with TCP packets, causing the router to drop legitimate packets, and forcing the retransmission of TCP packets. As such, an FB-Shrew attack can cause maximum damage with minimum resources. In this paper, we challenge an assumption of constant round trip time adopted in the original FB-Shrew model. As a result, this model fails to achieve its expected attack effect. In response, we analyze the TCP congestion window and queue behaviors to develop two low-high burst models for maximizing the potency of the FB-Shrew attack. Model 1 is designed to achieve the attack effect expected of the original model. Then, the attack potency of Model 1 is enhanced by simply adjusting the starting time of the attack burst to form Model 2. Mode 1 only exploits the retransmission timeout (RTO) mechanism. Model 2 takes advantage of both the RTO mechanism and the fast retransmission mechanism. In this way, Model 2 further slows down the growth of the congestion window and extends the attack period. A combination of theoretical analyses and simulations are adopted to first validate the proper functioning and effectiveness of the two models for a standard network configuration, and then we assess their attack performances with variations in different network parameters. Our performance assessment demonstrates that one attack unit of Model 2 damages almost twice the number of TCP units as one attack unit of Model 1, which represents an increase in attack potency of nearly 200 percent. The present study provides an expanded basis to explore FB-Shrew attack patterns that may be utilized by attackers. Moreover, the damage that could be inflicted by such attack and the extent to which defense strategies are capable of mitigating the attack's impact could be assessed more precisely by defenders.

Tuning the Aggressive TCP Behavior for Highly Concurrent HTTP Connections in Intra-Datacenter

Modern data centers host diverse hyper text transfer protocol (HTTP)-based services, which employ persistent transmission control protocol (TCP) connections to send HTTP requests and responses. However, the ON/OFF pattern of HTTP traffic disturbs the increase of TCP congestion window, potentially triggering packet loss at the beginning of ON period. Furthermore, the transmission performance becomes worse due to severe congestion in the concurrent transfer of HTTP response. In this paper, we provide the first extensive study to investigate the root cause of performance degradation of highly concurrent HTTP connections in data center network. We further present the design and implementation of TCP-TRIM, which employs probe packets to smooth the aggressive increase of congestion window in persistent TCP connection and leverages congestion detection and control at end-host to limit the growth of switch queue length under highly concurrent TCP connections. The experimental results of at-scale simulations and real implementations demonstrate that TCP-TRIM reduces the completion time of HTTP response by up to 80%, while introducing little deployment overhead only at the end hosts.

Analysis for TCP in data center networks: Outcast and Incast

The unfairness caused by bandwidth sharing via TCP in data center networks is called TCP Outcast problem. Some researchers show that the throughput of a flow with small Round Trip Time (RTT) is less than that with large RTT which is completely contrary to the classic Transmission Control Protocol (TCP) protocol, and they believe that the Outcast problem is caused by port blackout in data center. However, we find that there are more important reasons for outcast which are an unfair distribution of the flows with different RTT on physical link and the differences in congestion window size when finishing the transmission of current block. TCP Incast is a throughput collapse that occurs when many flows arrive at the switch destined for the same output port. Incast deteriorates the performance of networks by increasing the queuing delay of flows and decreasing the throughput of applications. In this paper, we develop an analytical model for goodput when TCP Outcast occurs. Then we propose a new protocol based on window notification mechanism called TCP congestion window replacement (TCP-CWR), which can improve the goodput of the flows with small RTT to solve the outcast problem. Second, we propose a new transport layer protocol based on acknowledgment (ACK) reply changing rate to solve the Incast problem named TCP with Acknowledgment Changing Rate (TCP-ACR). The TCP-ACR protocol adjusts the current congestion window according to the changing rate of ACK and the theoretical maximum congestion window, and it can alleviate the congestion in data center networks. Last, we conduct simulation experiments to verify our proposed schemes. The results show that our analysis of Outcast is correct and our proposed protocols for Outcast and Incast are effective and practical.

The Impact of TCP Congestion Window Size on the Performance Evaluation of Mobile Ad Hoc (Manet) Routing Protocols

A mobile ad hoc network (MANET) is a temporary collection of mobile nodes randomly moved within a limited terrain area. The nodes are connected to form a wireless network without use any communication infrastructure. Because of the limiting resources of MANET nodes, multiple hopsscheme is proposed for data exchange across the network. Varieties of mobile ad hoc routing protocols have been developed to support the multi-hop scheme of ad hoc networks. A popular Transmission Control Protocol (TCP) provides a reliable connection in a computer network environment; it sets its congestion window size in response to the behavior of the network to achieve the best performance. This work aims to investigate and compare the MANET protocolperformance, such as DSDV, AODV and DSR in terms of network throughput, average routing load, the packet delivery ratio (PDR), and average end-to-end delay by varying the maximum congestion window size. Our simulation has been implemented using a well-known NS-2.35 network simulator. The simulated results show that the demonstrates of the concepts of MANET routing protocols with respect to TCP congestion window size in MANET environment.

Fair and efficient Transmission Control Protocol access in the IEEE 802.11 infrastructure basic service set

AbstractWhen the stations in an IEEE 802.11 infrastructure basic service set employ Transmission Control Protocol (TCP), this exacerbates per‐flow unfair access problem. We propose a novel analytical model to approximately calculate the maximum per‐flow TCP congestion window limit that prevents packet losses at the access point buffer and therefore provides fair TCP access both in the downlink and uplink. The proposed analysis is unique in considering the effects of varying number of uplink and downlink TCP flows, differing round trip times among TCP connections and the use of delayed TCP acknowledgment (ACK) mechanism. Motivated by the findings of this theoretical analysis and simulations, we design a link layer access control block to be employed only at the access point in order to resolve the unfair access problem. The proposed link layer access control block uses congestion control and ACK filtering approach by prioritizing the access of TCP data packets of downlink flows over TCP ACK packets of uplink flows. Via simulations, we show that the proposed algorithm can provide both short‐term and long‐term fair accesses while improving channel utilization and access delay. Copyright © 2013 John Wiley & Sons, Ltd.

TCP NRT: a new TCP algorithm for differentiating non-congestion retransmission timeouts over multihop wireless networks

In multihop wireless networks, reliable data transfer is one of the most difficult tasks. When transmission control protocol (TCP) operates in multihop wireless networks, the performance of TCP reduces drastically. TCP retransmission timeouts (RTOs) related to non-congestion events such as spurious and random packet losses have been reported as one of the main problems in the performance degradation of TCP in multihop wireless networks. The RTOs triggered by random packet losses due to transmission errors lead to unnecessary reduction of TCP congestion window size, and the spurious RTOs due to sudden delay of packets on the network paths often cause unnecessary retransmissions as well as reduction of congestion window size. Existing solutions for detecting non-congestion RTOs have no mechanism to differentiate the spurious RTOs from RTOs caused by random packet loss. In this paper, we introduce an efficient algorithm called non-congestion retransmission timeouts (TCP NRT) which is capable of recovering packets after RTOs by reducing unnecessary retransmissions and needless reduction of congestion window size in order to improve the performance of TCP in multihop wireless networks. TCP NRT consists of three key components: NRT-detection, NRT-differentiation, and NRT-reaction. We implemented the algorithm in Qualnet network simulator and compared its performance to existing TCP versions. Results from the experiments show that our algorithm achieves significant performance improvement in terms of throughput and accuracy. Also, the results showed that our algorithm, TCP NRT, maintains a fair and friendly behavior compared to the most widely deployed TCP, NewReno.

A stochastic model for the behavior of multiple TCP NewReno sources over optical burst switching network

This work proposes a stochastic model to characterize the transmission control protocol (TCP) over optical burst switching (OBS) networks which helps to understand the interaction between the congestion control mechanism of TCP and the characteristic bursty losses in the OBS network. We derive the steady-state throughput of a TCP NewReno source by modeling it as a Markov chain and the OBS network as an open queueing network with rejection blocking. We model all the phases in the evolution of TCP congestion window and evaluate the number of packets sent and time spent in different states of TCP. We model the mixed assembly process, burst assembler and disassembler modules, and the core network using queueing theory and compute the burst loss probability and end-to-end delay in the network. We derive expression for the throughput of a TCP source by solving the models developed for the source and the network with a set of fixed-point equations. To evaluate the impact of a burst loss on each TCP flow accurately, we define the burst as a composition of per-flow-bursts (which is a burst of packets from a single source). Analytical and simulation results validate the model and highlight the importance of accounting for individual phases in the evolution of TCP congestion window.

TCP Congestion Window Adaptation Through Contention Detection in Ad Hoc Networks

In multihop ad hoc networks over 802.11, the real bandwidth-delay product (BDP) cannot reach a value as large as in wired networks due to contention through wireless shared channels. As a result, conventional congestion-control algorithms cause an overshooting window problem and result in poor throughput in ad hoc networks. In this paper, we split the real round-trip time (RTT) into two parts: 1) congestion RTT and 2) contention RTT. We reveal that the contention RTT has nothing to do with the BDP and that the BDP is determined by only the congestion RTT if a link with the worst contention status does not lead to link breakage. An inadequate use of contention RTT causes a Transmission Control Protocol (TCP) congestion window overshooting problem. We propose a novel mechanism called congestion window adaptation through contention detection, which provides a more accurate method of estimating the contention status. Based on this mechanism, we present a congestion window adaptation method to limit the window size from overshooting. Simulation results show that our proposed mechanism outperforms the conventional TCP and the enhanced mechanisms called TCP with contention control and TCP with a maximum window.

A TCP Driven CAC Scheme: Efficient Resource Utilization in a Leaky HAP-Satellite Integrated Scenario

An integrated high altitude platform (HAP)-satellite communication system appears to be very suitable for a large set of scenarios including emergency situations, exceptional events, etc. In fact, the satellite capability to provide a broadband and ubiquitous access can be enhanced by the deployment of HAP that allows the use of low-power consuming, cost-efficient, and portable terminals. To obtain an optimum utilization of radio resource, without renouncing to QoS satisfaction, a suitable call admission control scheme must be implemented. Nevertheless, transmission control protocol (TCP) behavior, mainly affected by the high latency and shadowing events, can impact call admission control (CAC) performance. Therefore, it would be desirable that the CAC scheme takes into account also the TCP congestion window real evolution. We present an innovative CAC scheme that uses TCP statistics as one of its inputs and is able to manage different classes of users. Results show that CAC performance is significantly improved by introducing TCP statistics about network congestion as an input parameter.