Estimation of packet losses due to propagation impairments: Application to TCP performance over satellite

The transmission control protocol (TCP) is widely used to provide reliable data transmission due to its congestion and flow control mechanisms that provide reliable error recovery in higher layers. In satellite links, various atmospheric phenomena may lead to high packet loss rate (PLR) degrading the TCP throughput. Modern satellite systems operate at frequencies above 10 GHz, where rainfall is the dominant fading mechanism leading to high bit error ratio and correlated packet losses. In this paper, a mathematical analysis is presented to accurately describe the statistical properties of the packet‐error process in a dynamically varying satellite channel. The proposed method is extended to provide PLR estimations when block forward error correction (FEC) is employed. A new Markov‐based method, based on the previous analysis and adapted to the rain‐faded satellite channel, is also presented for the estimation of TCP SACK throughput and tested against simulation results. Based on the information provided by the packet‐error model, a study between the TCP performance under various FEC schemes and a proposed adaptive FEC scheme has provided indications about the superiority of the proposed model. Copyright © 2007 John Wiley & Sons, Ltd.     

Long-Lived TCP Connections Via Satellite: Cross-Layer Bandwidth Allocation, Pricing, and Adaptive Control

The paper focuses on the assignment of a common bandwidth resource to TCP connections over a satellite channel. The connections are grouped according to their source–destination pairs, which correspond to the up- and down-link channels traversed, and each group may experience different fading conditions. By exploiting the tradeoff between bandwidth and channel redundancy (as determined by bit and coding rates) in the maximization of TCP goodput, an overall optimization problem is constructed, which can be solved by numerical techniques. Different relations between goodput maximization and fairness of the allocations are investigated, and a possible pricing scheme is proposed. The allocation strategies are tested and compared in a fading environment, first under static conditions, and then in a real dynamic scenario. The goodput-fairness optimization allows significant gains over bandwidth allocations only aimed at keeping the channel bit error rate below a given threshold in all fading conditions.     

Improving link layer performance on satellite channels with shadowing via delayed two-copy selective repeat ARQ

This paper focuses on improving performance of land mobile satellite channels (LMSCs) at high band (Ka-band or EHF band), where shadowing is the primary impediment to reliable data transmission. Compared with multipath fading, shadowing exists on a longer time scale; hence, interleaving to combat shadowing introduces unacceptably large decoding delay. We use Lutz's model to investigate bit-error rate/packet-error rate (BER/PER) performance of interleaving with various forward error correction (FEC) coding as a function of different channel parameters to demonstrate its limited effectiveness for combatting burst errors whose mean duration significantly exceed a link layer (LL) packet. We propose a delayed two-copy selective repeat ARQ (DTC-SR-ARQ) scheme, whereby two copies of a packet are sent-the second with a delay relative to the first-in every transmission or retransmission. Closed-form expressions for mean transmission time, success probability, and residual loss probability are provided and simulations used to validate the analysis. Furthermore, the issue of optimum delay is addressed as well, and a simple yet effective strategy is suggested to support transmission control protocol (TCP) traffic over this data link layer. DTC-SR-ARQ is shown to achieve much shorter additional delay than interleaving and compared with normal SR-ARQ, reduces mean transmission time at expense of a small increase in residual packet loss probability. Furthermore, ns2 simulation results show that for TCP traffic, DTC-SR-ARQ acquires higher end-to-end throughput than normal SR-ARQ.     

Modeling TCP SACK performance over wireless channels with semi-reliable ARQ/FEC

Providing reliable data communications over wireless channels is a challenging task because time-varying wireless channel characteristics often lead to bit errors. These errors result in loss of IP packets and, consequently, TCP segments encapsulated into these packets. Since TCP cannot distinguish packet losses due to bit corruption from those due to network congestion, any packet loss caused by wireless channel impairments leads to unnecessary execution of the TCP congestion control algorithms and, hence, sub-optimal performance. Automatic Repeat reQuest (ARQ) and Forward Error Correction (FEC) try to improve communication reliability and reduce packet losses by detecting and recovering corrupted bits. Most analytical models that studied the effect of ARQ and FEC on TCP performance assumed that the ARQ scheme is perfectly persistent (i.e., completely reliable), thus a frame is always successfully transmitted irrespective of the number of transmission attempts it takes. In this paper, we develop an analytical cross-layer model for a TCP connection running over a wireless channel with a semi-reliable ARQ scheme, where the amount of transmission attempts is limited by some number. The model allows to evaluate the joint effect of stochastic properties of the wireless channel characteristics and various implementation-specific parameters on TCP performance, which makes it suitable for performance optimization studies. The input parameters include the bit error rate, the value of the normalized autocorrelation function of bit error observations at lag 1, the strength of the FEC code, the persistency of ARQ, the size of protocol data units at different layers, the raw data rate of the wireless channel, and the bottleneck link buffer size.     

Suitability of DAMA and Contention-Based Satellite Access Schemes for TCP Traffic in Mobile DVB-RCS

The problem of optimizing access and bandwidth sharing among transmission control protocol (TCP) connections in the mobile digital video broadcasting return channel via satellite (DVB-RCS) is tackled in this paper. After sketching the general system architecture, we explicitly deal with the dynamic assignment of bandwidth to TCP connections on the return link, which is accomplished by a network control center (NCC) placed onboard the satellite. Mobile users access the satellite in multifrequency time-division multiple access (MF-TDMA), whereas they receive data from the NCC in time-division multiplexing (TDM). Two different techniques, based on deterministic and random access, are compared in terms of bandwidth usage and average completion time per connection, when the mobile user acts as both server and client. In the server case, to increase the TCP throughput, both packet-level forward error correction (FEC) on data sent by mobile users and a duplicated and delayed acknowledgment technique for TCP acknowledgment traffic from the NCC to the mobile users are applied. An analysis of the packet losses and a simulation campaign of file transfers by employing a realistic channel model has been carried out. The results of the analysis show the convenience of adopting a technique, in addition to the optimal data redundancy in different cases, such as the server or client role of users, their willingness to pay, the file size, and the environment type.     

A comparison of mechanisms for improving TCP performance overwireless links

Reliable transport protocols such as TCP are tuned to perform well in traditional networks where packet losses occur mostly because of congestion. However, networks with wireless and other lossy links also suffer from significant losses due to bit errors and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to end performance in wireless and lossy systems. We compare several schemes designed to improve the performance of TCP in such networks. We classify these schemes into three broad categories: end-to-end protocols, where loss recovery is performed by the sender; link-layer protocols that provide local reliability; and split-connection protocols that break the end-to-end connection into two parts at the base station. We present the results of several experiments performed in both LAN and WAN environments, using throughput and goodput as the metrics for comparison. Our results show that a reliable link-layer protocol that is TCP-aware provides very good performance. Furthermore, it is possible to achieve good performance without splitting the end-to-end connection at the base station. We also demonstrate that selective acknowledgments and explicit loss notifications result in significant performance improvements     

TCP-Jersey for wireless IP communications

Improving the performance of the transmission control protocol (TCP) in wireless Internet protocol (IP) communications has been an active research area. The performance degradation of TCP in wireless and wired-wireless hybrid networks is mainly due to its lack of the ability to differentiate the packet losses caused by network congestions from the losses caused by wireless link errors. In this paper, we propose a new TCP scheme, called TCP-Jersey, which is capable of distinguishing the wireless packet losses from the congestion packet losses, and reacting accordingly. TCP-Jersey consists of two key components, the available bandwidth estimation (ABE) algorithm and the congestion warning (CW) router configuration. ABE is a TCP sender side addition that continuously estimates the bandwidth available to the connection and guides the sender to adjust its transmission rate when the network becomes congested. CW is a configuration of network routers such that routers alert end stations by marking all packets when there is a sign of an incipient congestion. The marking of packets by the CW configured routers helps the sender of the TCP connection to effectively differentiate packet losses caused by network congestion from those caused by wireless link errors. This paper describes the design of TCP-Jersey, and presents results from experiments using the NS-2 network simulator. Results from simulations show that in a congestion free network with 1% of random wireless packet loss rate, TCP-Jersey achieves 17% and 85% improvements in goodput over TCP-Westwood and TCP-Reno, respectively; in a congested network where TCP flow competes with VoIP flows, with 1% of random wireless packet loss rate, TCP-Jersey achieves 9% and 76% improvements in goodput over TCP-Westwood and TCP-Reno, respectively. Our experiments of multiple TCP flows show that TCP-Jersey maintains the fair and friendly behavior with respect to other TCP flows.     

TCP Veno: TCP enhancement for transmission over wireless access networks

Wireless access networks in the form of wireless local area networks, home networks, and cellular networks are becoming an integral part of the Internet. Unlike wired networks, random packet loss due to bit errors is not negligible in wireless networks, and this causes significant performance degradation of transmission control protocol (TCP). We propose and study a novel end-to-end congestion control mechanism called TCP Veno that is simple and effective for dealing with random packet loss. A key ingredient of Veno is that it monitors the network congestion level and uses that information to decide whether packet losses are likely to be due to congestion or random bit errors. Specifically: (1) it refines the multiplicative decrease algorithm of TCP Reno-the most widely deployed TCP version in practice-by adjusting the slow-start threshold according to the perceived network congestion level rather than a fixed drop factor and (2) it refines the linear increase algorithm so that the connection can stay longer in an operating region in which the network bandwidth is fully utilized. Based on extensive network testbed experiments and live Internet measurements, we show that Veno can achieve significant throughput improvements without adversely affecting other concurrent TCP connections, including other concurrent Reno connections. In typical wireless access networks with 1% random packet loss rate, throughput improvement of up to 80% can be demonstrated. A salient feature of Veno is that it modifies only the sender-side protocol of Reno without changing the receiver-side protocol stack.     

Cross-layer-based adaptive TCP algorithm for cloud computing services in 4G LTE-A relaying communication

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.     

Storage area network extension solutions and their performance assessment

Several solutions are proposed to extend storage area networking solutions over distances of hundreds to thousands of kilometers. Native fibre channel or end-to-end fibre channel-based solutions can be offered over long distances using SONET-based networks. Protocols such as Internet SCSI (iSCSI), Internet fibre channel protocol (iFCP) and fibre channel over TCP/IP (FCIP) are being proposed to enable storage area networking solutions over networks that use IP as their transport protocol. Performance analysis of these solutions in terms of application throughput under variable network conditions of packet loss, bandwidth availability, extension distance, and TCP implementations in the sender and receiver is presented based on analytical modeling of different solutions.     

Rate-adaptive snoop: a TCP enhancement scheme over rate-controlled lossy links

We study TCP performance over the wireless links deploying a wireless rate-control technique, whose link characteristics are identified by variable link rate and bursty transmission error. We present a TCP enhancement scheme, called rate-adaptive snoop (RA-Snoop). RA-Snoop caches TCP packets selectively based on the wireless channel condition and the cached packets are retransmitted locally over the wireless link in case corruption loss is detected. In addition, for effective adaptation to variable bandwidth, RA-Snoop calculates the window feedback based on the bandwidth-delay product estimation and the queue level, then conveys this feedback information on the receiver's advertised window field in the acknowledgements returning to TCP sources. We compare the performance of RA-Snoop with that of existing schemes in the aspect of goodput and fairness. Results from simulations reveal that RA-Snoop achieves significant improvements over the existing schemes for various traffic scenarios.     

Modeling and analysis of WAP performance over wireless links

In this paper, an analytical model for studying the performance behavior of wireless application protocol (WAP) over wireless links is proposed. A Rayleigh fading channel model is used to characterize the behavior of the wireless channel. Mathematical expressions that represent the performance of WAP as a function of the protocol and the channel parameters are derived. Computer simulation results are presented to validate the analytical results. It is shown that WAP performs better in a bursty error environment than in a random error environment. The goodput of WAP can be improved by increasing the WAP packet group size, but the significance of the improvement depends on the underlying channel condition.     

卫星链路中TCP传输性能仿真与分析

卫星信道与地面有线信道具有不同的特点,当3TCP协议直接应用于卫星链路上时,其吞吐量等性能会受到影响。针对卫星链路的特点,文中分析了卫星链路中TCP协议存在的主要问题,使用NS2仿真软件,建立了仿真模型,并在给定的仿真环境下得出了一些结论。卫星链路具有时延大、误码率高等特点,仿真结果说明了当数据大小一定时,比特误码率越高,系统吞吐量越低,引起的丢包率越高;时延（往返时间）越长,系统吞吐量越低。最后,分析比较了优化TCP的运行参数、修改TCP控制机制、分割连接技术等卫星链路TCP协议的主要解决方案。     

ATCP: TCP for mobile ad hoc networks

Transport connections set up in wireless ad hoc networks are plagued by problems such as high bit error rates, frequent route changes, and partitions. If we run the transmission control protocol (TCP) over such connections, the throughput of the connection is observed to be extremely poor because TCP treats lost or delayed acknowledgments as congestion. We present an approach where we implement a thin layer between Internet protocol and standard TCP that corrects these problems and maintains high end-to-end TCP throughput. We have implemented our protocol in FreeBSD, and we present results from extensive experimentation done in an ad hoc network. We show that our solution improves the TCP's throughput by a factor of 2-3     

Reliable transmission of high-quality video over ATM networks

The development of broadband networks has led to the possibility of a wide variety of new and improved service offerings. Packetized video is likely to be one of the most significant high-bandwidth users of such networks. The transmission of variable bit-rate (VBR) video offers the potential promise of constant video quality but is generally accompanied by packet loss which significantly diminishes this potential. We study a class of error recovery schemes employing forward error-control (FEC) coding to recover from such losses. In particular, we show that a hybrid error recovery strategy involving the use of active FEC in tandem with simple passive error concealment schemes offers very robust performance even under high packet losses. We discuss two different methods of applying FEC to alleviate the problem of packet loss. The conventional method of applying FEC generally allocates additional bandwidth for channel coding while maintaining a specified average video coding rate. Such an approach suffers performance degradations at high loads since the bandwidth expansion associated with the use of FEC creates additional congestion that negates the potential benefit in using FEC. In contrast, we study a more efficient FEC application technique in our hybrid approach, which allocates bandwidth for channel coding by throttling the source coder rate (i.e., performing higher compression) while maintaining a fixed overall transmission rate. More specifically, we consider the performance of the hybrid approach where the bandwidth to accommodate the FEC overhead is made available by throttling the source coder rate sufficiently so that the overall rate after application of FEC is identical to that of the original unprotected system. We obtain the operational rate-distortion characteristics of such a scheme employing selected FEC codes. In doing so, we demonstrate the robust performance achieved by appropriate use of FEC under moderate-to-high packet losses in comparison to the unprotected system.     

Quantitative analysis of restricted reliability protocols for constant rate traffic

Restricted reliability, or limited reliability, transport layer protocols reduce but do not eliminate the packet loss probability seen by higher layers. Forward error correction (FEC) is employed to control the packet loss rate, but automatic repeat request (ARQ) is not used, either because no reverse channel exists or because it does not support the required quality of service. This paper analyzes the effect of FEC on packet loss probability and protocol delay and jitter. We demonstrate the tradeoff between parameters, showing how increased network traffic can be traded against reduced delay and jitter. The theoretical results are confirmed by simulation.     

Transport protocols for Internet-compatible satellite networks

We address the question of how well end-to-end transport connections perform in a satellite environment composed of one or more satellites in geostationary orbit (GEO) or low-altitude Earth orbit (LEO), in which the connection may traverse a portion of the wired Internet. We first summarize the various ways in which latency and asymmetry can impair the performance of the Internet's transmission control protocol (TCP), and discuss extensions to standard TCP that alleviate some of these performance problems. Through analysis, simulation, and experiments, we quantify the performance of state-of-the-art TCP implementations in a satellite environment. A key part of the experimental method is the use of traffic models empirically derived from Internet traffic traces. We identify those TCP implementations that can be expected to perform reasonably well, and those that can suffer serious performance degradation. An important result is that, even with the best satellite-optimized TCP implementations, moderate levels of congestion in the wide-area Internet can seriously degrade performance for satellite connections. For scenarios in which TCP performance is poor, we investigate the potential improvement of using a satellite gateway, proxy, or Web cache to “split” transport connections in a manner transparent to end users. Finally, we describe a new transport protocol for use internally within a satellite network or as part of a split connection. This protocol, which we call the satellite transport protocol (STP), is optimized for challenging network impairments such as high latency, asymmetry, and high error rates. Among its chief benefits are up to an order of magnitude reduction in the bandwidth used in the reverse path, as compared to standard TCP, when conducting large file transfers. This is a particularly important attribute for the kind of asymmetric connectivity likely to dominate satellite-based Internet access     

Rate-distortion optimized hybrid error control for real-time packetized video transmission.

The problem of application-layer error control for real-time video transmission over packet lossy networks is commonly addressed via joint source-channel coding (JSCC), where source coding and forward error correction (FEC) are jointly designed to compensate for packet losses. In this paper, we consider hybrid application-layer error correction consisting of FEC and retransmissions. The study is carried out in an integrated joint source-channel coding (IJSCC) framework, where error resilient source coding, channel coding, and error concealment are jointly considered in order to achieve the best video delivery quality. We first show the advantage of the proposed IJSCC framework as compared to a sequential JSCC approach, where error resilient source coding and channel coding are not fully integrated. In the USCC framework, we also study the performance of different error control scenarios, such as pure FEC, pure retransmission, and their combination. Pure FEC and application layer retransmissions are shown to each achieve optimal results depending on the packet loss rates and the round-trip time. A hybrid of FEC and retransmissions is shown to outperform each component individually due to its greater flexibility.     

A bit error rate analysis for TCP traffic over parallel free space photonics

Inter-satellite links (ISL) are a useful technology to transmit data to space stations and to communicate between satellites. However, there are serious limitations due to long delays and poor channel performance, resulting in high bit error rates (BER). In this paper, parallel transmission and the scaling of the Transport Control Protocol (TCP) window in free space optics (FSO) communications are analyzed in order to overcome these disadvantages in optical inter-satellite links. Latency and BER are the dominant effects that determine link performance. Thus, a physical, link, network and transport cross-layer analysis for FSO over ISL is presented in this paper. This analysis shows the advantages and disadvantages of using optical parallel transmission and TCP window scaling for free space optical links between stations and satellite constellations. The key contribution of this work is to simulate the effects of the BER and to link the results to packet error rate (PER) to determine the goodput for TCP transmissions by using a cross-layering approach. The results give evidence that wavelength division multiplexing (WDM) can mitigate the effects of long delay and high BER for a FSO communication using TCP.