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User-space protocols deliver high performance to applications on a low-cost Gb/s LAN 用户空间协议在低成本Gb/s LAN上为应用程序提供高性能
Aled Edwards, G. Watson, J. Lumley, D. Banks, Costas Calamvokis, Chris R. Dalton
Two important questions in high-speed networking are firstly, how to provide Gbit/s networking at low cost and secondly, how to provide a flexible low-level network interface so that applications can control their data from the instant it arrives. We describe some work that addresses both of these questions. The Jetstream Gbit/s LAN is an experimental, low-cost network interface that provides the services required by delay-sensitive traffic as well as meeting the performance needs of current applications. Jetstream is a combination of traditional shared-medium LAN technology and more recent ATM cell- and switch-based technology. Jetstream frames contain a channel identifier so that the network driver can immediately associate an incoming frame with its application. We have developed such a driver that enables applications to control how their data should be managed without the need to first move the data into the application's address space. Consequently, applications can elect to read just a part of a frame and then instruct the driver to move the remainder directly to its destination. Individual channels can elect to receive frames that have failed their CRC, while applications can specify frame-drop policies on a per-channel basis. Measured results show that both kernel- and user-space protocols can achieve very good throughput: applications using both TCP and our own reliable byte-stream protocol have demonstrated throughputs in excess of 200 Mbit/s. The benefits of running protocols in user-space are well known- the drawback has often been a severe penalty in the performance achieved. In this paper we show that it is possible to have the best of both worlds.
高速网络面临的两个重要问题是:如何以低成本提供Gbit/s的网络;如何提供灵活的底层网络接口,使应用程序能够从数据到达的那一刻起对其进行控制。我们描述了一些解决这两个问题的工作。Jetstream Gbit/s LAN是一种实验性的低成本网络接口,可提供对延迟敏感的流量所需的服务,并满足当前应用的性能需求。Jetstream是传统的共享介质局域网技术与最近的基于ATM单元和交换机的技术的结合。射流帧包含一个通道标识符,以便网络驱动程序可以立即将传入帧与其应用程序关联起来。我们已经开发了这样一个驱动程序,使应用程序能够控制如何管理它们的数据,而不需要首先将数据移动到应用程序的地址空间中。因此,应用程序可以选择只读取帧的一部分,然后指示驱动程序将其余部分直接移动到目的地。各个通道可以选择接收CRC失败的帧,而应用程序可以在每个通道的基础上指定丢帧策略。测量结果表明,内核空间和用户空间协议都可以实现非常好的吞吐量:使用TCP和我们自己的可靠字节流协议的应用程序已经证明吞吐量超过200 Mbit/s。在用户空间中运行协议的好处是众所周知的——缺点往往是性能上的严重损失。在本文中,我们证明了两全其美是可能的。
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引用次数: 91
TCP Vegas: new techniques for congestion detection and avoidance TCP Vegas:拥塞检测和避免的新技术
L. Brakmo, S. O'Malley, L. Peterson
Vegas is a new implementation of TCP that achieves between 40 and 70% better throughput, with one-fifth to one-half the losses, as compared to the implementation of TCP in the Reno distribution of BSD Unix. This paper motivates and describes the three key techniques employed by Vegas, and presents the results of a comprehensive experimental performance study—using both simulations and measurements on the Internet—of the Vegas and Reno implementations of TCP.
Vegas是一种新的TCP实现,与BSD Unix的Reno发行版中的TCP实现相比,它的吞吐量提高了40%到70%,损失减少了五分之一到一半。本文提出并描述了Vegas采用的三种关键技术,并介绍了一项综合实验性能研究的结果-使用互联网上的模拟和测量- Vegas和Reno TCP的实现。
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引用次数: 1606
Credit-based flow control for ATM networks: credit update protocol, adaptive credit allocation and statistical multiplexing 基于信用的ATM网络流量控制:信用更新协议、自适应信用分配和统计复用
H. T. Kung, T. Blackwell, A. Chapman
This paper presents three new results concerning credit-based flow control for ATM networks: (1) a simple and robust credit update protocol (CUP) suited for relatively inexpensive hardware/software implementation; (2) automatic adaptation of credit buffer allocation for virtual circuits (VCs) sharing the same buffer pool; (3) use of credit-based flow control to improve the effectiveness of statistical multiplexing in minimizing switch memory. These results have been substantiated by analysis, simulation and implementation.
本文提出了ATM网络中基于信用的流量控制的三个新成果:(1)一种简单且鲁棒的信用更新协议(CUP),适用于相对廉价的硬件/软件实现;(2)共享同一缓冲池的虚拟电路(vc)信用缓冲分配自动自适应;(3)利用基于信用的流量控制来提高统计复用的有效性,使交换机内存最小化。这些结果已通过分析、仿真和实现得到证实。
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引用次数: 235
An algorithm for lossless smoothing of MPEG video 一种MPEG视频无损平滑算法
S. Lam, Simon Chow, David K. Y. Yau
Interframe compression techniques, such as those used in MPEG video, give rise to a coded bit stream where picture sizes differ by a factor of 10 or more. As a result, buffering is needed to reduce (smooth) rate fluctuations of encoder output from one picture to the next; without smoothing, the performance of networks that carry such video traffic would be adversely affected. Various techniques have been suggested for controlling the output rate of a VBR encoder to alleviate network congestion or prevent buffer overflow. Most of these techniques, however, are lossy, and should be used only as a last resort. In this paper, we design and specify an algorithm for lossless smoothing. The algorithm is characterized by three parameters: D (delay bound), K (number of pictures with known sizes), and H (lookahead interval). We present a theorem which guarantees that, if K ≥ 1, the algorithm finds a solution that satisfies the delay bound. (Although the algorithm and theorem were motivated by MPEG video, they are applicable to the smoothing of compressed video in general). To study performance characteristics of the algorithm, we conducted a large number of experiments using statistics from four MPEG video sequences.
帧间压缩技术,例如在MPEG视频中使用的技术,产生编码的比特流,其中图像大小相差10倍或更多。因此,需要缓冲来减少编码器输出从一张图片到下一张图片的(平滑)速率波动;如果没有平滑,承载此类视频流量的网络的性能将受到不利影响。为了缓解网络拥塞或防止缓冲区溢出,已经提出了各种控制VBR编码器输出速率的技术。然而,这些技术中的大多数都是有损的,只能作为最后的手段使用。本文设计并实现了一种无损平滑算法。该算法的特征是三个参数:D(延迟界),K(已知大小的图片数量)和H(向前看间隔)。我们给出了一个定理,保证当K≥1时,算法找到一个满足延迟界的解。(虽然算法和定理是由MPEG视频激发的,但它们一般适用于压缩视频的平滑处理)。为了研究该算法的性能特征,我们利用四个MPEG视频序列的统计数据进行了大量的实验。
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引用次数: 151
USC: a universal stub compiler USC:通用存根编译器
S. O'Malley, Todd A. Proebsting, Allen Brady Montz
USC is a new stub compiler that generates stubs that perform many data conversion operations. USC is flexible and can be used in situations where previously only manual code generation was possible. USC generated code is up to 20 times faster than code generated by traditional argument marshaling schemes such as ASN.1 and Sun XDR. This paper presents the design of USC and a comprehensive set of experiments that compares USC performance with the best manually generated code and traditional stub compilers.
USC是一种新的存根编译器,它生成执行许多数据转换操作的存根。USC是灵活的,可以在以前只能手工生成代码的情况下使用。USC生成的代码比传统参数封送方案(如ASN.1和Sun XDR)生成的代码快20倍。本文介绍了USC的设计和一组全面的实验,将USC的性能与最佳的手动生成代码和传统的存根编译器进行比较。
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引用次数: 76
Scalable feedback control for multicast video distribution in the Internet 互联网多播视频分发的可伸缩反馈控制
J. Bolot, T. Turletti, I. Wakeman
We describe a mechanism for scalable control of multicast continuous media streams. The mechanism uses a novel probing mechanism to solicit feedback information in a scalable manner and to estimate the number of receivers. In addition, it separates the congestion signal from the congestion control algorithm, so as to cope with heterogeneous networks. This mechanism has been implemented in the IVS video conference system using options within RTP to elicit information about the quality of the video delivered to the receivers. The H.261 coder of IVS then uses this information to adjust its output rate, the goal being to maximize the perceptual quality of the image received at the destinations while minimizing the bandwidth used by the video transmission. We find that our prototype control mechanism is well suited to the Internet environment. Furthermore, it prevents video sources from creating congestion in the Internet. Experiments are underway to investigate how the scalable proving mechanism can be used to facilitate multicast video distribution to large number of participants.
我们描述了一种可扩展控制多播连续媒体流的机制。该机制采用一种新颖的探测机制,以可扩展的方式征求反馈信息,并估计接收器的数量。此外,它将拥塞信号从拥塞控制算法中分离出来,以应对异构网络。这一机制已在IVS视频会议系统中实施,使用RTP内的选项来获取有关传送给接收者的视频质量的信息。IVS的H.261编码器然后使用这些信息来调整其输出速率,目标是最大限度地提高目的地接收到的图像的感知质量,同时最大限度地减少视频传输使用的带宽。我们发现我们的原型控制机制非常适合互联网环境。此外,它还可以防止视频源在互联网上造成拥塞。实验正在进行中,以研究如何使用可扩展的证明机制来促进多播视频分发给大量参与者。
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引用次数: 429
On optimization of polling policy represented by neural network 基于神经网络的轮询策略优化研究
Y. Matsumoto
This paper deals with the problem of scheduling a server in a polling system with multiple queues and complete information. We represent the polling policy by a neural network; namely, given the number of waiting customers in each queue, the server determines next queue he should visit according to the output of the neural network. By using the simulated annealing method, we improve the neural polling policy in such a way that the mean delay of customers is minimized. Numerical results show that the present approach is especially valid for asymmetric polling systems whose analytical optimization is considered intractable.
研究了具有完整信息的多队列轮询系统中的服务器调度问题。我们用神经网络表示轮询策略;即,给定每个队列中等待的客户数量,服务器根据神经网络的输出确定下一个应该访问的队列。通过模拟退火方法,改进了神经轮询策略,使客户的平均延迟最小。数值结果表明,该方法特别适用于难以进行解析优化的非对称轮询系统。
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引用次数: 6
An architecture for wide-area multicast routing 广域组播路由体系结构
S. Deering, D. Estrin, D. Farinacci, V. Jacobson, Ching-Gung Liu, Liming Wei
Existing multicast routing mechanisms were intended for use within regions where a group is widely represented or bandwidth is universally plentiful. When group members, and senders to those group members, are distributed sparsely across a wide area, these schemes are not efficient; data packets or membership report information are occasionally sent over many links that do not lead to receivers or senders, respectively. We have developed a multicast routing architecture that efficiently establishes distribution trees across wide area internets, where many groups will be sparsely represented. Efficiency is measured in terms of the state, control message processing, and data packet processing, required across the entire network in order to deliver data packets to the members of the group. Our Protocol Independent Multicast (PIM) architecture: (a) maintains the traditional IP multicast service model of receiver-initiated membership; (b) can be configured to adapt to different multicast group and network characteristics; (c) is not dependent on a specific unicast routing protocol; and (d) uses soft-state mechanisms to adapt to underlying network conditions and group dynamics. The robustness, flexibility, and scaling properties of this architecture make it well suited to large heterogeneous inter-networks.
现有的组播路由机制是为了在组广泛代表或带宽普遍充足的区域内使用。当群组成员和发送者分布在广泛的区域时,这些方案是不有效的;数据包或成员报告信息偶尔会通过多个链路发送,这些链路分别不指向接收方或发送方。我们已经开发了一种多播路由架构,它可以在广域互联网上有效地建立分布树,在广域互联网上,许多组将被稀疏地表示。效率是根据状态、控制消息处理和数据包处理来衡量的,为了将数据包传递给组的成员,需要在整个网络中进行这些处理。我们的协议独立组播(PIM)架构:(a)保持传统的IP组播服务模型的接收方发起的成员;(b)可配置以适应不同的组播组和网络特性;(c)不依赖于特定的单播路由协议;(d)使用软状态机制来适应潜在的网络条件和群体动态。该体系结构的健壮性、灵活性和可伸缩性使其非常适合大型异构互连网络。
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引用次数: 465
Distributed, scalable routing based on link-state vectors 基于链路状态向量的分布式、可扩展路由
Jochen Behrens, J. Garcia-Luna-Aceves
A new family of routing algorithms for the distributed maintenance of routing information in large networks and internets is introduced. This family is called link vector algorithms (LVA), and is based on the selective diffusion of link-state information based on the distributed computation of preferred paths, rather than on the flooding of complete link-state information based on the distributed computation of preferred paths, rather than on the flooding of complete link-state information to all routers. According to LVA, each router maintains a subset of the topology that corresponds to the links used by its neighbor routers in their preferred paths to known destinations. Based on that subset of topology information, the router derives its own preferred paths and communicates the corresponding link-state information to its neighbors. An update message contains a vector of updates; each such update specifies a link and its parameters. LVAs can be used for different types of routing. The correctness of LVA is verified for arbitrary types of routing when correct and deterministic algorithms are used to select preferred paths at each router. LVA is shown to have smaller complexity than link-state and distance-vector algorithms, and to have better average performance than the ideal topology-broadcast algorithm and the distributed Bellman-Ford algorithm.
介绍了一种新的路由算法,用于大型网络和互联网中路由信息的分布式维护。这类算法被称为链路向量算法(link vector algorithms, LVA),它基于基于首选路径的分布式计算的链路状态信息的选择性扩散,而不是基于基于首选路径的分布式计算的完整链路状态信息的泛洪,也不是基于将完整链路状态信息泛洪到所有路由器。根据LVA,每个路由器维护一个拓扑的子集,该子集对应于其邻居路由器在到达已知目的地的首选路径中使用的链路。路由器根据拓扑信息子集派生出自己的首选路径,并将相应的链路状态信息传递给邻居。更新消息包含更新向量;每个这样的更新都指定一个链接及其参数。lva可用于不同类型的路由。当在每个路由器上使用正确的、确定性的算法选择优选路径时,验证了任意类型路由的LVA的正确性。LVA算法比链路状态算法和距离矢量算法具有更小的复杂度,比理想拓扑广播算法和分布式Bellman-Ford算法具有更好的平均性能。
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引用次数: 69
A methodology for designing communication protocols 设计通信协议的方法学
Gurdip Singh
We propose a compositional technique for designing protocols. The technique involves specifying constraints between the events of the component protocols. A constraint may either require synchronization between certain events of the component protocols or may require inhibiting an event in one protocol on the occurrence of an event in another component protocol. We find both types of constraints useful in composing protocols. We demonstrate the applicability of the technique by deriving several protocols. The technique facilitates modular design and verification. Our technique, in conjunction with the sequential composition technique, can be used to design complex protocols.
我们提出了一种组合技术来设计协议。该技术涉及在组件协议的事件之间指定约束。约束可以要求在组件协议的某些事件之间进行同步,也可以要求在一个协议中的事件发生时抑制另一个组件协议中的事件。我们发现这两种类型的约束在组合协议时都很有用。我们通过推导几个协议来证明该技术的适用性。该技术有利于模块化设计和验证。我们的技术,结合顺序组合技术,可以用来设计复杂的协议。
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引用次数: 6
期刊
Proceedings of the conference on Communications architectures, protocols and applications
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