Retransmission Timeout Research Articles

The taming of the shrew: mitigating low-rate TCP-targeted attack

A Shrew attack, which uses a low-rate burst carefully designed to exploit TCP's retransmission timeout mechanism, can throttle the bandwidth of a TCP flow in a stealthy manner. While such an attack can significantly degrade the performance of all TCP-based protocols and services including Internet routing (e.g., BGP), no existing scheme clearly solves the problem in real network scenarios. In this paper, we propose a simple protection mechanism, called SAP (Shrew Attack Protection), for defending against a Shrew attack. Rather than attempting to track and isolate Shrew attackers, SAP identifies TCP victims by monitoring their drop rates and preferentially admits those packets from the victims with high drop rates to the output queue. This is to ensure that well-behaved TCP sessions can retain their bandwidth shares. Our simulation results indicate that under a Shrew attack, SAP can prevent TCP sessions from closing, and effectively enable TCP flows to maintain high throughput. SAP is a destination-port-based mechanism and requires only a small number of counters to find potential victims, which makes SAP readily implementable on top of existing router mechanisms.

Analysis of RTO Caused by Retransmission Loss to Combat Channel Noise

A Retransmission Time Out (RTO) is inevitable, when the retransmission of a packet fails to reach the receiver. An RTO compels TCP to reduce packet flow drastically. However, in case of an RTO resulting from retransmission failure caused by the channel noise, reduction in the flow is inappropriate. The problem is compounded when a TCP sender is forbidden to continue transmission till the occurrence of the timeout. In this paper, we investigate the impact of such RTOs with the help of an empirical mathematical analysis. The analysis presented in the paper calculates the idle period of the sender in terms of number of RTTs, which depends on the value of congestion window before the timeout. The mathematical analysis is supported by the results of simulation based experiments and the evaluation of a scheme that improves TCP performance in case of avoidable timeouts caused by the loss of retransmission on an erroneous wireless link.

Improving TCP performance over mobile ad hoc networks

The transmission control protocol (TCP) was designed to provide reliable end-to-end delivery of data over unreliable networks. Most TCP implementations have been carefully optimised to operate in wired networks. Deploying standard TCP in wireless mobile ad hoc networks (MANETs) can lead to poor performance of TCP. The main problem that faces TCP in ad hoc networks is its inability to distinguish between packet losses due to congestion and those due to route failures. In order to adapt TCP to MANET environment, several TCP improvements that differentiate between different types of packet losses using explicit notification messages from lower layers have been proposed. In this paper, we propose a new end-to-end technique that does not require any participation from intermediate nodes within the network. The proposed scheme relies on expanding the retransmission timeout (RTO) interval in order to reduce the number of timeouts that do not result from congestion. Simulation results show that our scheme can improve TCP throughput and delay. It outperformed the standard TCP in detailed simulation experiments.

Detecting TCP Retransmission Timeouts Non-related to Congestion in Multi-Hop Wireless Networks

Detecting TCP Retransmission Timeouts Non-related to Congestion in Multi-Hop Wireless Networks

NQAR: Network Quality Aware Routing in Error-Prone Wireless Sensor Networks

We propose a network quality aware routing (NQAR) mechanism to provide an enabling method of the delay-sensitive data delivery over error-prone wireless sensor networks. Unlike the existing routing methods that select routes with the shortest arrival latency or the minimum hop count, the proposed scheme adaptively selects the route based on the network qualities including link errors and collisions with minimum additional complexity. It is designed to avoid the paths with potential noise and collision that may cause many non-deterministic backoffs and retransmissions. We propose a generic framework to select a minimum cost route that takes the packet loss rate and collision history into account. NQAR uses a data centric approach to estimate a single-hop delay based on processing time, propagation delay, packet loss rate, number of backoffs, and the retransmission timeout between two neighboring nodes. This enables a source node to choose the shortest expected end-to-end delay path to send a delay-sensitive data. The experiment results show that NQAR reduces the end-to-end transfer delay up to approximately 50% in comparison with the latency-based directed diffusion and the hop count-based directed diffusion under the error-prone network environments. Moreover, NQAR shows better performance than those routing methods in terms of jitter, reachability, and network lifetime.

Two schemes to reduce latency in short lived TCP flows

Motivated by the amount of short-lived TCP traffic due to interactive applications like Web browsing, we propose two schemes to reduce the download time of TCP flows. Both schemes explicitly try to overcome the long delay suffered when TCP has to recover from the loss of the last data segment in a flow, loss for which fast retransmit cannot be triggered. The first scheme consists in transmitting twice the last segment, while the second one exploits the reception of the duplicate ACK caused by the delivery of the FIN segment. Both schemes allow the transmitter to recover the loss without waiting for a retransmission timeout expiration, thus providing quicker recovery with up to 50% gain.

Tuning SCTP failover for carrier grade telephony signaling

The Stream Control Transmission Protocol (SCTP) has not only been selected as the signaling transport protocol of choice in IETF SIGTRAN, the architecture that bridges circuit-switched and IP-based mobile core networks, but also plays a pivotal role in SAE/LTE, the next-generation UMTS/HSPA networks. To meet the redundancy requirements of telecom signaling traffic, SCTP includes a failover mechanism that enables rerouting of traffic from an unreachable network path to a backup path. However, the recommendations provided by IETF on how to configure the SCTP failover mechanism to meet telecom signaling requirements are kept quite general and leave much of the tuning to the telecom equipment vendor and/or operator. Several works by us and others have been carried out to study the effect of different SCTP parameters on the failover performance. The main contribution of this paper is that it gives a coherent treatment of how to configure the SCTP failover mechanism for carrier-grade telephony signaling, and provides practically usable configuration recommendations. The paper also discusses an alternate or complementary way of optimizing the SCTP failover mechanism by relaxing the exponential backoff that foregoes a retransmission timeout in SCTP. Some results showing significantly reduced failover times by use of this mechanism, with only marginal deteriorating effects on a signaling network, are discussed and analyzed in the paper.

Safe and effective fine-grained TCP retransmissions for datacenter communication

This paper presents a practical solution to a problem facing high-fan-in, high-bandwidth synchronized TCP workloads in datacenter Ethernets---the TCP incast problem. In these networks, receivers can experience a drastic reduction in application throughput when simultaneously requesting data from many servers using TCP. Inbound data overfills small switch buffers, leading to TCP timeouts lasting hundreds of milliseconds. For many datacenter workloads that have a barrier synchronization requirement (e.g., filesystem reads and parallel data-intensive queries), throughput is reduced by up to 90%. For latency-sensitive applications, TCP timeouts in the datacenter impose delays of hundreds of milliseconds in networks with round-trip-times in microseconds. Our practical solution uses high-resolution timers to enable microsecond-granularity TCP timeouts. We demonstrate that this technique is effective in avoiding TCP incast collapse in simulation and in real-world experiments. We show that eliminating the minimum retransmission timeout bound is safe for all environments, including the wide-area.

Performance enhancement of mSCTP for vertical handover across heterogeneous wireless networks

AbstractIn the mobile Stream Control Transmission Protocol (mSCTP) for vertical handover, a mobile user may suffer from performance degradation due to the problems of packet reordering and retransmission timeout and due to the packet loss during handover. To solve these problems, we propose a new scheme of handover retransmission for mSCTP handover, in which the correspondent node retransmits the outstanding data packets to the mobile node over the new primary path. From simulation results, it is shown that the proposed scheme can avoid the packet reordering and retransmission timeout problems during handover. Moreover, we can see that the proposed scheme can significantly improve throughput of mSCTP handover, compared with the existing schemes. Copyright © 2009 John Wiley & Sons, Ltd.

On the properties of an adaptive TCP Minimum RTO

The Internet Engineering Task Force (IETF) specified in RFC 2988 the Minimum TCP–RTO and recommended that the TCP Retransmission Timeout (RTO) should not be smaller than 1 s. According to RFC 2988, there are two main limitations that call for a lower bound to protect TCP from spurious timeouts: (i) the OS clock granularity ( 500 ms for most OSs at the time of RFC 2988 publication) and (ii) the Delayed Acknowledgments (usually set to 200 ms ). We evaluate the correctness of the suggested policy and investigate the impact of the conservative 1-s Minimum TCP–RTO under modern networking conditions. We define a Cost Function to capture the impact of the extra waiting time, due to the conservative IETF specification. Our experimental analysis reveals that the OS clock granularity should not be a matter of concern for modern OSs; we carefully design a mechanism to deal with Delayed ACKs. We call the proposed mechanism Adaptive MINRTO (AdMINRTO) to reflect its operational properties. In particular, the mechanism identifies the packets whose ACKs are (possibly) going to be delayed and applies extended Minimum RTO to these packets only, in order to avoid spurious timeout expirations; otherwise, the Minimum RTO is adjusted to smaller values in order to avoid extensive idle periods. We show, through simulations, that the Adaptive MINRTO improves TCP performance significantly, especially in case of next generation’s high-speed, wireless-access networks. The impact of the proposed mechanism, performance-wise, depends on several network conditions. For example, the impact of the Adaptive MINRTO increases (i) with the Packet Error Rate (PER), (ii) when the path Round-Trip Time is short and (iii) for fast transmission links (i.e., high-speeds).

NJ+: An Efficient Congestion Control Mechanism for Wireless Networks

Transmission control protocols have to overcome common problems in wireless networks. TCP employing both packet loss discrimination mechanism and available bandwidth estimation algorithm, known as the good existing solution, shows significant performance enhancement in wireless networks. For instance, TCP New Jersey which exhibits high throughput in wireless networks intends to improve TCP performance by using available bandwidth estimation and congestion warning. Even though it achieves 17% and 85% improvements in terms of goodput over TCP Westwood and TCP Reno, respectively, we further improve it by exploring maximized available bandwidth estimation, handling bit-error-rate error recovery, and effective adjustment of sending rate for retransmission timeout. Hence, we propose TCP NJ+, showing that for up to 5% packet loss rate, it outperforms other TCP variants by 19% to 104% in terms of goodput when the network is in bi-directional background traffic.

Performance Metrics for IEEE 802.21 Media Independent Handover (MIH) Signaling

The IEEE 802.21 Media Independent Handover (MIH) working group is developing a set of mechanisms to facilitate migration of mobile users between access networks that use different link-layer technologies. Among these are mobility managers that create and process signaling messages to facilitate handovers. The MIH signaling architecture being developed in the Internet Engineering Task Force (IETF) allows any transport layer protocol to carry MIH messages. The IETF has considered using the unreliable but lightweight transport available with the User Datagram Protocol (UDP) as well as the reliable stream-oriented transport with congestion control offered by the Transmission Control Protocol (TCP). In this paper we develop mathematical models that result in expressions for the characteristic function of the time required to complete exchanges of an arbitrary number of MIH signaling messages between a mobile node (MN) and a remote mobility manager (MM). Our models also provide expressions for the average amount of overhead associated with MIH message exchanges due to retransmissions either by the MIH signaling entities or by the transport-layer protocol. In addition, we provide simulation results that confirm the results from the mathematical model and illustrate the effect of varying transport parameters such as the TCP maximum retransmission timeout.

Link Layer Adaptation for Shared Wireless Links

While traditional link layer protocols assume that they fully control the underlying link, in contemporary wireless networks the link may be dynamically shared by sessions belonging to different users and/or applications. To assess the impact of link sharing, we measure the File Transfer and Web Browsing throughput achieved over a Selective Repeat (SR) protocol, with or without contention from Media Distribution. Our results indicate that the optimal protocol settings strongly depend on the level of contention for the link. We therefore present two link layer protocols that adapt to the available bandwidth, our Adaptive Selective Repeat (ASR) protocol which dynamically modifies its retransmission timeouts, and the Radio Link Control (RLC) protocol specified for use by Universal Mobile Telecommunications System networks which does not employ retransmission timers. We first repeat our performance measurements to determine the optimal settings for each protocol, and then compare the fine tuned versions of all protocols with respect to their File Transfer and Web Browsing throughput, as well as to the delay induced to the contending Media Distribution packets. Our results indicate that while both RLC and ASR are more stable than SR, the complex RLC does not match the performance of our simpler ASR.

Robust Cross-Layer Design of Wireless-Profiled TCP Mobile Receiver for Vertical Handover

In this paper, we consider downward and upward vertical handovers in integrated wireless LAN and cellular networks and address wireless-profiled Transmission Control Protocol premature timeouts due to step increase of round-trip time (RTT) and false fast retransmit due to packet reordering. Specifically, we develop a mobile-receiver centric loosely coupled cross-layer design, which is easy to implement and deploy, backward compatible with the Wireless Application Protocol version 2 architecture, and robust in the absence of perfect cross-layer information. We propose two proactive schemes, called RTT Inflation and RTT Equalization. The RTT Inflation scheme suppresses premature timeouts by carefully inflating retransmission timeout time, and the RTT Equalization scheme prevents false fast retransmit by equalizing the round-trip delay experienced by all packets. We conduct extensive simulations to evaluate the performance in downward and upward vertical handovers. It is demonstrated that the proposed schemes significantly improve the performance in a wide range of network conditions.

Interleaved Traffic Splitting: A promising technique to solve False Timeout

In this paper, we analyze the False Timeout (FTO) problem that TCP flows suffered in OBS networks and propose Interleaved Traffic Splitting (ITS) to solve the problem. We show that the collision loss of ACK bursts may also cause FTO, which has been neglected by previous researchers. As TCP Sack is designed to recover from multiple packet losses, we find that a Sack flow can recover its data transmission without triggering retransmission timeout if packet loss is not too severe. This suggests limiting the number of packets carried by a single burst. In ITS, this is achieved by carrying out a parallel, interleaved burst assembly process at each ingress OBS edge router. We show that ITS tackles the FTO problem caused by the loss of both data and ACK bursts. Unlike other existing approaches, the ITS scheme requires only modification to the assembly/de-assembly mechanisms implemented at the edge routers. It does not add complexities to the network management or the router design. In this paper, both analytical and simulation results are presented to demonstrate the performance improvement of ITS. Our analysis further shows that the use of two parallel bursts is optimum for ITS.

Design of coordinator to eliminate redundant local retransmission in wireless networks

Owing to limited bandwidth, high bit error rate, and bursty error in the wireless environment, the performance of the transmission control protocol (TCP) degrades greatly in wireless networks. Up to now, many researchers have contributed greatly to the wireless TCP field. However, in most of their works, the wireless TCP module usually works in the TCP layer and has no idea of the actual time of the packet transmission, which is determined by the Scheduler in the media access control (MAC) layer, and this will bring the inaccuracy to the local retransmission timeout and induce the redundant local retransmission. In this article, a coordinator is introduced into the base-station (BS), which can provide efficient cooperation between the TCP module and the scheduler module. On the bais of the performance analysis and simulation results, the proposed method is shown to eliminate redundant local retransmission, increase throughput, and improve TCP-level fairness in wireless networks. Moreover, this scheme is orthogonal to those existing wireless TCP schemes, thus it can give great compatibility to the current networks, and further enhance the performance of TCP under the condition that the performance improvement benefiting from the existing approaches will not be affected.

Limitations of Equation-Based Congestion Control

We study limitations of an equation-based congestion control protocol, called TCP-friendly rate control (TFRC). It examines how the three main factors that determine TFRC throughput, namely, the TCP-friendly equation, loss event rate estimation, and delay estimation, can influence the long-term throughput imbalance between TFRC and TCP. Especially, we show that different sending rates of competing flows cause these flows to experience different loss event rates. There are several fundamental reasons why TFRC and TCP flows have different average sending rates, from the first place. Earlier work shows that the convexity of the TCP-friendly equation used in TFRC causes the sending rate difference. We report two additional reasons in this paper: 1) the convexity of where is a loss event period and 2) different retransmission timeout period (RTO) estimations of TCP and TFRC. These factors can be the reasons for TCP and TFRC to experience initially different sending rates. But we find that the loss event rate difference due to the differing sending rates greatly amplifies the initial throughput difference; in some extreme cases, TFRC uses around 20 times more, or sometimes 10 times less, bandwidth than TCP. Despite these factors influencing the throughput difference, we also find that simple heuristics can greatly mitigate the problem.

An Adaptive Media-Aware Retransmission Timeout Estimation Method for Low-Delay Packet Video

Time-constrained error recovery is an integral component of reliable low-delay video applications. Regardless of the error-control method adopted by the application, unacknowledged or missing packets must be quickly identified as lost or delayed, so that necessary actions can be taken by the server/client on time. Historically, this problem has been referred to as retransmission timeout (RTO) estimation. Earlier studies show that existing RTO estimators suffer from either long loss detection times or a large number of spurious timeouts. The goal of this study is to address these problems by developing an RTO estimation method specifically tailored for low-delay video applications. In the media-unaware mode, this method exploits the temporal dependence in packet delay to optimally manage the tradeoff between the amount of overwaiting and redundant retransmission rate. As opposed to existing methods, our approach is completely adaptive to the source video characteristics and time-varying network conditions, and does not use any preset parameters. In the media-aware mode, on the other hand, the timeout estimates are jointly optimized based on the importance and urgency of the video packets such that the rendering quality is maximized under the given rate constraints. With a comprehensive set of simulation and experimental results, we show that both the media-unaware and media-aware RTO estimators detect lost packets faster and more accurately than their rivals. Furthermore, our results also substantiate the fact that the media-aware RTO estimator outperforms all other RTO estimators in terms of video quality

Why TCP timers (still) don’t work well

We argue that the design principles of the TCP timeout algorithm are based solely on RTT estimations and may lead to flow synchronization, unnecessary retransmission effort and unfair resource allocation. We present a new Window-Based Retransmission Timeout algorithm (WB-RTO) for TCP, which exhibits two major properties: (i) it cancels retransmission synchronization, which dominates when resource demand exceeds resource supply and (ii) it reschedules flows on the basis of their contribution to congestion. WB-RTO achieves better fairness and slightly better goodput with significant less retransmission effort.

Reverse acknowledgment (ra) scheme: fast detection of retransmission failures for SR-ARQ

This letter proposes a simple modification called reverse acknowledgement (RA) scheme to a negative acknowledgement (NAK)-based selective repeat automatic repeat request (SR-ARQ) protocol for its fast loss recovery. By adopting the RA scheme, a NAK-based SR-ARQ receiver can detect and recover retransmission failures without invoking retransmission timeouts (RTOs). By OPNET simulations, we evaluate the performance of the RA scheme in terms of the transport delay and re-sequencing delay and show that the RA scheme considerably improves the performance of NAK-based SR-ARQ.