Analysis of TCP Performance over Mobile Ad Hoc Networks

Mobile ad hoc networks have attracted attention lately as a means of providing continuous network connectivity to mobile computing devices regardless of physical location. Recent research has focused primarily on the routing protocols needed in such an environment. In this paper, we investigate the effects that link breakage due to mobility has on TCP performance. Through simulation, we show that TCP throughput drops significantly when nodes move, due to TCP's inability to recognize the difference between link failure and congestion. We also analyze specific examples, such as a situation where throughput is zero for a particular connection. We introduce a new metric, expected throughput, for the comparison of throughput in multi-hop networks, and then use this metric to show how the use of explicit link failure notification (ELFN) techniques can significantly improve TCP performance.     

无线自组织网络TCP容量优化

为了计算无线自组织网络中准确的TCP容量,并利用其进行有效的窗口控制,以避免拥塞窗口增长过速的问题,提出一种适用于无线自组织网络TCP容量的优化计算。此方法基于时延和空间复用率,考虑网络中竞争干扰,解决了以往的计算方法对拓扑不能广泛适用的问题。模拟结果显示,基于文章的TCP容量进行拥塞窗口调节,能有效地提高网络的吞吐量。     

无线Ad Hoc网中的TCP SACK与TCP Vegas

本文用仿真方法分析了TCP SACK和Vegas在无线ab hoc网络中的性能，提出了一种改进的SACK选项格式（ASACK）和一种称为环回时间通知（RN）的新技术以分别用于提高TCP SACK和TCP Vegas在无线ab hoc网中的性能。为了研究路由稳定性TCP Vegas性能的影响，我们实现了一种基于相关性的选路协议（ABR）。     

Multicasting with Localized Control in Wireless Ad Hoc Networks

This paper investigates how to support multicasting in wireless ad hoc networks without throttling the dominant unicast flows. Unicast flows are usually congestion-controlled with protocols like TCP. However, there are no such protocols for multicast flows in wireless ad hoc networks and multicast flows can therefore cause severe congestion and throttle TCP-like flows in these environments. Based on a cross-layer approach, this paper proposes a completely-localized scheme to prevent multicast flows from causing severe congestion and the associated deleterious effects on other flows in wireless ad hoc networks. The proposed scheme combines the layered multicast concept with the routing-based congestion avoidance idea to reduce the aggregated rate of multicast flows when they use excessive bandwidth on a wireless link. Our analysis and extensive simulations show that the fully-localized scheme proposed in this paper is effective in ensuring the fairness of bandwidth sharing between multicast and unicast flows in wireless ad hoc networks.     

大带宽时延积网络TCP Vegas自适应慢启动算法

TCP Vegas具有比TCP Reno更好的带宽利用能力和稳定性,但是在带宽时延积较大的网络中,TCP Vegas会出现慢启动过早结束、拥塞窗口过小的问题,降低了传输效率.文中在分析慢启动结束的原因和条件的基础上,提出一种对临时性排队时延进行估计,将其排除后再进行慢启动结束条件判断的TCP Vegas慢启动算法,对不同网络条件有自适应能力.仿真结果表明本算法能有效避免慢启动过早结束,使TCP性能明显改善.     

Cross-layer TCP with bitmap error recovery scheme in wireless ad hoc networks

Current TCP is not able to distinguish corruption losses from packet loss events. Hence, high transmission errors and varying inherent latency within a wireless network would cause seriously adverse effects to TCP performance. To improve TCP in IEEE 802.11 multi-hop ad hoc wireless networks, this study proposes an error recovery mechanism based on coordination of TCP and IEEE 802.11 MAC protocols. The simulation results confirm that the proposed error recovery approach could provide a more efficient solution for frequent transmission losses, and enable TCP to distinguish between congestion errors and transmission errors, and thus, to respond with proper remedial actions.     

A transport protocol for supporting multimedia streaming in mobile ad hoc networks

Transport protocol design for supporting multimedia streaming in mobile ad hoc networks is challenging because of unique issues, including mobility-induced disconnection, reconnection, and high out-of-order delivery ratios; channel errors and network congestion. In this paper, we describe the design and implementation of a transmission control protocol (TCP)-friendly transport protocol for ad hoc networks. Our key design novelty is to perform multimetric joint identification for packet and connection behaviors based on end-to-end measurements. Our NS-2 simulations show significant performance improvement over wired TCP friendly congestion control and TCP with explicit-link-failure-notification support in ad hoc networks.     

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     

在Ad Hoc网络中TCP-SACK性能研究及改进

在Ad Hoc网中,经常发生链路失效和路由变化。TCP把数据包的丢失归结为链路拥塞,所以TCP在Ad Hoc网中表现很差。在本文中通过应用一种新的方法,基于传输包乱序检测与响应方法,研究TCP在Ad Hoc网中的性能。通过实验结果表明,此算法解决了,提高了网络的吞吐量。     

DTPA: A Reliable Datagram Transport Protocol over Ad Hoc Networks

As a prevalent reliable transport protocol in the Internet, TCP uses two key functions: AIMD (Additive Increase Multiplicative Decrease) congestion control and cumulative ACK technique to guarantee delivery. However, with these two functions, TCP becomes lowly efficient in ad hoc networks that have a much lower BDP and frequent packet losses due to various reasons, since TCP adjusts its transmission window based on packet losses. In this paper, we present that, provided that the BDP is very small, any AIMD-style congestion control is costly and hence not necessary for ad hoc networks. On the contrary, a technique to guarantee reliable transmission and to recover packet losses plays a more critical role in the design of a transport protocol over ad hoc networks. With this basis, we propose a novel and effective datagram-oriented end-to-end reliable transport protocol for ad hoc networks, which we call DTPA. The proposed scheme incorporates a fixed-size window based flow control and a cumulative bit-vector based selective ACK strategy. A mathematical model is developed to evaluate the performance of DTPA and to determine the optimum transmission window used in DTPA. The protocol is verified using GloMoSim. The simulation results show that our proposal substantially improves the network performance.     

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.     

Throughput and delay analysis for hybrid radio-frequency and free-space-optical (RF/FSO) networks

In this paper the per-node throughput and end-to-end delay of randomly deployed (i.e. ad-hoc) hybrid radio frequency - free space optics (RF/FSO) networks are studied. The hybrid RF/FSO network consists of an RF ad hoc network of n nodes, f(n) of them, termed ‘super nodes’, are equipped with an additional FSO transceiver with transmission range s(n). Every RF and FSO transceiver is able to transmit at a maximum data rate of W ₁ and W ₂ bits/sec, respectively. An upper bound on the per node throughput capacity is derived. In order to prove that this upper bound is achievable, a hybrid routing scheme is designed whereby the data traffic is divided into two classes and assigned different forwarding strategies. The capacity improvement with the support of FSO nodes is evaluated and compared against the corresponding results for pure RF wireless networks. Under optimal throughput scaling, the scaling of average end-to-end delay is derived. A significant gain in throughput capacity and a notable reduction in delay will be achieved if \(f(n) = \Upomega\left(\frac{1}{s(n)}\sqrt{\frac{n}{\log n}}\cdot \frac{W_1}{W_2} \right)\). Furthermore, it is found that for fixed W ₁, f(n) and n where f(n) < n, there is no capacity incentive to increase the FSO data rate beyond a critical value. In addition, both throughput and delay can achieve linear scaling by properly adjusting the FSO transmission range and the number of FSO nodes.     

An enhanced congestion avoidance mechanism for TCP Vegas

TCP Vegas detects network congestion in the early stage and successfully prevents periodic packet loss that usually occurs in traditional schemes. It has been demonstrated that TCP Vegas achieves much higher throughput than TCP Reno. However, TCP Vegas cannot prevent unnecessary throughput degradation when congestion occurs in the backward path. In this letter, we propose an enhanced congestion avoidance mechanism for TCP Vegas. By distinguishing whether congestion occurs in the forward path or not, it significantly improves the connection throughput when the backward path is congested.     

TCP CERL: congestion control enhancement over wireless networks

In this paper, we propose and verify a modified version of TCP Reno that we call TCP Congestion Control Enhancement for Random Loss (CERL). We compare the performance of TCP CERL, using simulations conducted in ns-2, to the following other TCP variants: TCP Reno, TCP NewReno, TCP Vegas, TCP WestwoodNR and TCP Veno. TCP CERL is a sender-side modification of TCP Reno. It improves the performance of TCP in wireless networks subject to random losses. It utilizes the RTT measurements made throughout the duration of the connection to estimate the queue length of the link, and then estimates the congestion status. By distinguishing random losses from congestion losses based on a dynamically set threshold value, TCP CERL successfully attacks the well-known performance degradation issue of TCP over channels subject to random losses. Unlike other TCP variants, TCP CERL doesn’t reduce the congestion window and slow start threshold when random loss is detected. It is very simple to implement, yet provides a significant throughput gain over the other TCP variants mentioned above. In single connection tests, TCP CERL achieved an 175, 153, 85, 64 and 88% throughput gain over TCP Reno, TCP NewReno, TCP Vegas, TCP WestwoodNR and TCP Veno, respectively. In tests with multiple coexisting connections, TCP CERL achieved an 211, 226, 123, 70 and 199% throughput improvement over TCP Reno, TCP NewReno, TCP Vegas, TCP WestwoodNR and TCP Veno, respectively.     

CAPACITY EVALUATION OF MULTI-CHANNEL WIRELESS AD HOC NETWORKS

In this paper, the capacity of multi-channel, multi-hop ad hoc network is evaluated.In particular, the performance of multi-hop ad hoc network with single channel IEEE 802.11 MAC utilizing different topologies is shown. Also the scaling laws of throughputs for largescale ad hoc networks and the theoretical guaranteed throughput bounds for multi-channel grid topology systems are proposed. The results presented in this work will help researchers to choose the proper parameter settings in evaluation of protocols for multi-hop ad hoc networks.     

无线自组网与Internet互连中TCP流公平性分析与改进

通过实验与分析揭示了基于IEEE 802.11 MAC的无线自组网与Internet互连时TCP流间的不公平性现象及其产生原因,并基于网关带宽占用对互连环境下TCP流间的公平性进行定义,提出了一种基于网关动态带宽控制的TCP流公平性解决方案TCP-GAFC。TCP-GAFC只在网关节点上进行实现,无需修改TCP协议和MAC协议,具有很好的适用性和可扩展性。仿真实验表明,TCP-GAFC显著地改进了互连环境下TCP流间的公平性,并能达到合理的总吞吐量。     

Auction-Based Bandwidth Allocation in Multi-Hop Wireless Ad Hoc Networks

In this paper, we propose a multi-hop auction-based bandwidth allocation mechanism to address the flow contention problem in wireless ad hoc networks. By modeling the problem as an iterative auction-based structure, it enables us to derive fair and efficient bandwidth allocation to each node on the basis of only local information. Further, a multi-hop flow coordination mechanism is then developed to optimize the network performance. Simulation results suggest that the proposed mechanism outperforms other approaches in terms of network throughput, bandwidth utilization, fairness, end-to-end delay, packet loss rate, and robustness.     

On throughput capacity of large-scale ad hoc networks with realistic buffer constraint

The problem of determining the throughput capacity of an ad hoc network is addressed. Previous studies mainly focused on the infinite buffer scenario, however, in this paper we consider a large-scale ad hoc network with a scalable traffic model, where each node has a buffer of size B packets, and explore its corresponding per node throughput performance. We first model each node as a G/G/1/B queuing system which incorporates the important wireless interference and medium access contention. With the help of this queuing model, we then explore the properties of the throughput upper bound for all scheduling schemes. Based on these properties, we further develop an analytical approach to derive the expressions of per node throughput capacity for the concerned buffer-limited ad hoc network. The results show that the cumulative effect of packet loss due to the per hop buffer overflowing will degrade the throughput performance, and the degradation is inversely proportional to the buffer size. Finally, we provide the specific scheduling schemes which enable the per node throughput to approach its upper bound, under both symmetrical and unsymmetrical network topologies.     

Distributed Approximation Algorithms for Spectrum Allocation in Wireless ad Hoc Networks

Spectrum allocation is a difficult and hot issue in wireless ad hoc networks. An efficient method of spectrum allocation is a key factor to improve quality of service and performance of wireless networks. In this paper, we consider the spectrum allocation problem which asks how to allocate the least number of spectrum blocks in a field to ensure the service on any random k locations simultaneously. Our solution to the spectrum allocation problem is the minimum k-Roman dominating set. We propose two distributed algorithms for the issue of spectrum allocation in wireless ad hoc networks. One is a distributed 6k-approximation algorithm for the spectrum allocation of satisfying any random k (k ≥ 2) locations in the class of unit ball graphs. The other one is a better distributed algorithm for finding a (1 + ε)-approximation for the spectrum allocation problem of serving any random two locations, in the class of growth-bounded graphs. We also describe the simulation results and analyze them.     

Simulation-based analysis of a multi-hop integrated UMTS and WLAN network

The next generation communication networks will comprise of third-generation (3G) and fourth-generation (4G) cellular and multi-hop ad hoc networks. In particular, the Universal Mobile Telecommunications System (UMTS) and the IEEE 802.11 ad hoc networks will play a significant role. The UMTS can provide Internet service in wide areas with excellent mobility support while IEEE 802.11 can provide high speed last hop connectivity in indoor environment. The performance of the offered end-to-end TCP connection when these two different technologies come together to serve the users is an important aspect to be studied. In this paper, we build an analytical model to model the complete process of TCP packet transfer over this integrated network. The delay and end-to-end throughput are modeled. We also verify the simulation results considering the usual error rate in UMTS channels. Our analytical results match well with the simulation models that are widely used. Apart from modeling the Internet connectivity, our model can be used for multitude of tasks, such as gateway selection, resource reservation, etc., in the next generation cellular multi-hop networks.