Cause Of Packet Loss Research Articles

Robust end-to-end loss differentiation scheme for transport control protocol over wired/wireless networks

The authors propose a robust end-to-end loss differentiation scheme to identify the packet losses because of congestion for transport control protocol (TCP) connections over wired/wireless networks. The authors use the measured round trip time (RTT) values to determine whether the cause of packet loss is because of the congestion over wired path or regular bit errors over wireless paths. The classification should be as accurate as possible to achieve high throughput and maximum fairness for the TCP connections sharing the wired/wireless paths. The accuracies of previous schemes in the literature depends on varying network parameters such as RTT, buffer size, amount of cross traffic, wireless loss rate and congestion loss rate. The proposed scheme is robust in that the accuracy remains rather stable under varying network parameters. The basic idea behind the scheme is to set the threshold for the classification to be a function of the minimum RTT and the current sample RTT, so that it may automatically adapt itself to the current congestion level. When the congestion level of the path is estimated to be low, the threshold for a packet loss to be classified as a congestion loss is increased. This avoids unnecessary halving of the congestion window on packet loss because of the regular bit errors over the wireless path and hence improves the TCP throughput. When the congestion level of the path is estimated to be high, the threshold for a packet loss to be classified as the congestion loss not to miss any congestion loss is decreased and hence improves the TCP fairness. In ns 2 simulations, the proposed scheme correctly classifies the congestion losses under varying network parameters whereas the previous schemes show some dependency on subsets of parameters.

Cross-layer loss differentiation algorithms to improve TCP performance in WLANs

Loss Differentiation Algorithms (LDA) are currently used to determine the cause of packet losses with an aim of improving TCP performance over wireless networks. In this work, we propose a cross-layer solution using two complementary LDA schemes in order to classify the loss origin on an 802.11 link and then to react consequently. The first LDA scheme, acting at the MAC layer, allows differentiating losses due to signal failure caused by displacement or by noise from other loss types. Moreover, in the case of a signal failure, this scheme adapts the behavior of the MAC layer in order to avoid a costly end-to-end TCP resolution. The objective of the second LDA scheme, which acts at the TCP layer, is to distinguish between losses due to interferences and those due to congestions, then accordingly adapt the TCP behavior. We finally demonstrate, through simulation, the efficiency of each LDA scheme as well as the whole cross-layer solution.

Rate-Based Supervisory Congestion Control for Ad Hoc Networks

In this letter, we suggest a rate-based supervisory congestion control scheme for the ad hoc networks that use TCP as the transport protocol. This scheme makes it possible for the TCP sender to distinguish the causes of packet loss. In addition, this scheme guarantees the fair sharing of the available bandwidth among the connections. We show the reliability of our scheme by using the supervisory control framework and simulations confirm the effectiveness of our scheme.

A survey of tcp enhancements for last-hop wireless networks

TCP continues to be an important transport-layer communication protocol that is typically tuned to perform well in traditional wired networks, where bit error rate (BER) is low and congestion is the primary cause of packet loss. The emergence of various mobile access networks has prompted researchers to look for suitable modifications to TCP so as to make it survive in the wireless era as well. Networks with wireless links and mobile hosts incur significant losses due to high BER, host motion, and handoff mechanisms. Mobile devices face temporary and unannounced loss of network connectivity when they move. They are likely to have scarce resources, and they react to frequent changes in the environment. Motion causes varying, increased delays and packet losses, while the network learns how to deliver packets to the new location of the host. TCP incorrectly interprets these delays and losses as signs of network congestion and throttles its transmission rate, causing degraded end-to-end performance. This article provides an in-depth survey of various TCP enhancements which are specifically targeted for last-hop wireless environments. The objective is to review the performance issues of TCP variations, when employed in last-hop wireless networks, and to provide a categorized analysis of different existing solutions comparatively, as we all know that it is difficult to create a “one size fits all” TCP for last-hop wireless networks.

Snug-Vegas and Snug-Reno: efficient mechanisms for performance improvement of TCP over heterogeneous networks

Improving the performance of the traditional TCP in wireless IP communications has been an active research area. The significant cause of packet losses in such heterogenous networks is no longer limited to network congestion. The performance degradation of TCP in wireless and wired-wireless hybrid networks is mainly due to its lack of ability to differentiate the packet losses caused by network congestions from the losses caused by wireless link errors. New variants of TCP Vegas and TCP Reno named Snug-Vegas and Snug-Reno, respectively, are proposed. By using random-loss indications marked by base stations, Snug-Vegas and Snug-Reno may detect random packet losses precisely. Through the packet loss differentiation, Snug-Vegas and Snug-Reno react appropriately to the losses, and based on the simulation results, it is seen that the throughput of connection over heterogeneous networks can be significantly improved.

Rate control for streaming video over wireless

Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is TCP-friendly rate control (TFRC) (Floyd, 2000). It is equation-based rate control in which the TCP-friendly rate is determined as a function of packet loss rate, round-trip time, and packet size. TFRC assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the main cause of packet loss is at the physical layer. In this article we review existing approaches to solve this problem. Then we propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, but also results in full utilization of the wireless channel. NS-2 simulations, actual experiments over a 1/spl times/RTT CDMA wireless data network, and video streaming simulations using traces from the actual experiments are carried out to characterize the performance and show the efficiency of our proposed approach.

Improving TCP Performance in Mobile Networks

In this letter, a new transport layer mechanism is proposed to improve the performance of transport control protocol (TCP) in mobile networks. The proposed mechanism is comprised of two parts: a loss classifier (LC) and a congestion window extrapolator (CWE). Based on LC, the cause of packet loss during roaming is determined. If the loss is considered to be caused by congestion in the wireline, the congestion window is halved; otherwise, the packet is considered to be lost in the last hop, the wireless portion, and the sender adjusts the size of the congestion window based on CWE. We conduct simulations to evaluate the performance of the proposed mechanism. The results show that our mechanism significantly improves TCP performance as compared with existing solutions for mobile networks.

TCP in wireless environments: problems and solutions

The Internet provides a platform for rapid and timely information exchange among a disparate array of clients and servers. TCP and IP are separately designed and closely tied protocols that define the rules of communication between end hosts, and are the most commonly used protocol suite for data transfer in the Internet. The combination of TCP/IP dominates today's communication in various networks from the wired backbone to the heterogeneous network due to its remarkable simplicity and reliability, TCP has become the de facto standard used in most applications ranging from interactive sessions such as Telnet and HTTP, to bulk data transfer like FTP. TCP was originally designed primarily for wired networks. In a wired network, random bit error rate, a characteristic usually more pronounced in the wireless network, is negligible, and congestion is the main cause of packet loss. The emerging wireless applications, especially high-speed multimedia services and the advent of wireless IP communications carried by the Internet, call for calibration and sophisticated enhancement or modifications of this protocol suite for improved performance. Based on the assumption that packet losses are signals of network congestion, the additive increase multiplicative decrease congestion control of the standard TCP protocol reaches the steady state, which reflects the protocol's efficiency in terms of throughput and link utilization. However, this assumption does not hold when the end-to-end path also includes wireless links. Factors such as high BER, unstable channel characteristics, and user mobility may all contribute to packet losses. Many studies have shown that the unmodified standard TCP performs poorly in a wireless environment due to its inability to distinguish packet losses caused by network congestion from those attributed to transmission errors. In this article, following a brief introduction to TCP, we analyze the problems TCP exhibits in the wireless IP communication environment, and illustrate viable solutions by detailed examples.

Link-level measurements from an 802.11b mesh network

This paper analyzes the causes of packet loss in a 38-node urban multi-hop 802.11b network. The patterns and causes of loss are important in the design of routing and error-correction protocols, as well as in network planning.The paper makes the following observations. The distribution of inter-node loss rates is relatively uniform over the whole range of loss rates; there is no clear threshold separating "in range" and "out of range." Most links have relatively stable loss rates from one second to the next, though a small minority have very bursty losses at that time scale. Signal-to-noise ratio and distance have little predictive value for loss rate. The large number of links with intermediate loss rates is probably due to multi-path fading rather than attenuation or interference.The phenomena discussed here are all well-known. The contributions of this paper are an understanding of their relative importance, of how they interact, and of the implications for MAC and routing protocol design.

Explicit transport error notification (ETEN) for error-prone wireless and satellite networks

Wireless and satellite networks often have non-negligible packet corruption rates that can significantly degrade TCP performance. This is due to TCP’s assumption that every packet loss is an indication of network congestion (causing TCP to reduce the transmission rate). This problem has received much attention in the literature. In this paper, we take a broad look at the problem of enhancing TCP performance under corruption losses, and include a discussion of the key issues. The main contributions of this paper are: (i) a confirmation of previous studies that show the reduction of TCP performance in the face of corruption loss, and in addition a plausible upper bound achievable with perfect knowledge of the cause of loss, (ii) a classification of the potential mitigation space, and (iii) the introduction of a promising new mitigation that employs rich cumulative information from intermediate nodes in a path to form a better congestion response. We first illustrate the performance implications of corruption-based loss for a variety of networks via simulation. In addition, we show a rough upper bound on the performance gains a TCP could get if it could perfectly determine the cause of each segment loss––independent of any specific mechanism for TCP to learn the root cause of packet loss. Next, we provide a taxonomy of potential practical classes of mitigations that TCP end-points and intermediate network elements can cooperatively use to decrease the performance impact of corruption-based loss. Finally, we briefly consider a potential mitigation, called cumulative explicit transport error notification (CETEN), which covers a portion of the solution space previously unexplored. CETEN is shown to be a promising mitigation strategy, but a strategy with numerous formidable practical hurdles still to overcome.

TCP Westwood with adaptive bandwidth estimation to improve efficiency/friendliness tradeoffs

In this paper, we propose an extension of Transmission Control Protocol (TCP) Westwood allowing the management of the Efficiency/Friendliness-to-NewReno tradeoffs. We show that the extended TCP Westwood is able to achieve higher total link utilization, yet at the same time maintain friendliness. TCP Westwood (for short, TCPW) implements a novel window congestion control algorithm based on eligible rate estimation (RE). The performance of TCPW has been promising, exceeding that of TCP NewReno in ‘large leaky pipes’; i.e. network paths with high bandwidth-delay product and non-negligible random error rate. Consider the situation where TCPW and TCP NewReno connections coexist and share common bottlenecks. Friendliness in this shared environment is paramount. Under certain conditions TCP NewReno may experience some performance degradation since TCPW ‘learns’ more about connection performance and thus can take better advantage of available bandwidth. To manage the efficiency/friendliness tradeoffs, we propose to combine the original TCPW Bandwidth Estimation (BE) strategy with a new RE strategy. One finds that BE provides significantly higher utilization, but may, under certain conditions, overestimates a connection fair share. RE, on the other hand, tends to be closer to the achieved rate of a connection, but it may underestimate the connection fair share. The question is: which estimate—RE or BE—yields better throughput/friendliness tradeoffs? Our studies show that RE works best when packet loss is mostly due to congestion. If, on the other hand, packet loss is mostly due to link errors, BE gives better performance. To achieve the ‘best of all worlds’, we introduce a method we call Combined Rate and Bandwidth estimation (CRB.) A connection first infers the predominant cause of packet loss (buffer congestion or random error) and then uses the more appropriate estimation method. Simulation shows that the adaptive CRB provides a very effective compromise between efficiency and friendliness.

Improving reliable transport and handoff performance in cellular wireless networks

TCP is a reliable transport protocol tuned to perform well in traditional networks where congestion is the primary cause of packet loss. However, networks with wireless links and mobile hosts incur significant losses due to bit-errors and hand-offs. This environment violates many of the assumptions made by TCP, causing degraded end-to-end performance. In this paper, we describe the additions and modifications to the standard Internet protocol stack (TCP/IP) to improve end-to-end reliable transport performance in mobile environments. The protocol changes are made to network-layer software at the base station and mobile host, and preserve the end-to-end semantics of TCP. One part of the modifications, called the snoop module, caches packets at the base station and performs local retransmissions across the wireless link to alleviate the problems caused by high bit-error rates. The second part is a routing protocol that enables low-latency handoff to occur with negligible data loss. We have implemented this new protocol stack on a wireless testbed. Our experiments show that this system is significantly more robust at dealing with unreliable wireless links than normal TCP; we have achieved throughput speedups of up to 20 times over regular TCP and handoff latencies over 10 times shorter than other mobile routing protocols.