A High Throughput Packet-Switched Network Technique Without Message Reassembly

A new packet-switching network technique is described which, while utilizing certain aspects of the ARPANET technology, introduces a substantially different technique for handling traffic which is longer than a single packet in length. The technique is keyed to a common-user network environment, where a wide variety of subscriber types, ranging from computers to simple terminals, are to be serviced. Subscribers in most cases would be remotely located from the network switching nodes. By splitting the buffering between the originating and destination nodes and by essentially eliminating the segment reassembly process, substantial reductions in on-line buffering can be achieved, while still maintaining short response times for interactive messages and large bandwidths for long data exchanges. In this paper we describe the network operational concepts and traffic flow for various subscriber types, show specific examples and timing diagrams for message flows, and present a comparative analysis of the buffer sizing, throughput, and delay for this new technique compared to the well-known ARPANET technique of packet switching.     

Robust throughput and routing for mobile ad hoc wireless networks

Flows transported across mobile ad hoc wireless networks suffer from route breakups caused by nodal mobility. In a network that aims to support critical interactive real-time data transactions, to provide for the uninterrupted execution of a transaction, or for the rapid transport of a high value file, it is essential to identify robust routes across which such transactions are transported. Noting that route failures can induce long re-routing delays that may be highly interruptive for many applications and message/stream transactions, it is beneficial to configure the routing scheme to send a flow across a route whose lifetime is longer, with sufficiently high probability, than the estimated duration of the activity that it is selected to carry. We evaluate the ability of a mobile ad hoc wireless network to distribute flows across robust routes by introducing the robust throughput measure as a performance metric. The utility gained by the delivery of flow messages is based on the level of interruption experienced by the underlying transaction. As a special case, for certain applications only transactions that are completed without being prematurely interrupted may convey data to their intended users that is of acceptable utility. We describe the mathematical calculation of a network’s robust throughput measure, as well as its robust throughput capacity. We introduce the robust flow admission and routing algorithm (RFAR) to provide for the timely and robust transport of flow transactions across mobile ad hoc wireless network systems.     

Hybrid mobile backbone network routing with flow control and distance awareness (MBNR-FC/DA)

The mobile backbone network (MBN) architecture has been introduced to synthesize robust, scalable and efficient mobile ad hoc wireless networks that support multimedia flows. Backbone capable nodes are dynamically elected to construct a mobile backbone (Bnet). In this article, we present a hybrid routing mechanism for such networks, identified as MBN routing with flow control and distance awareness (MBNR-FC/DA) scheme. Flows that travel a distance longer than a threshold level are routed across the Bnet. This induces a significant reduction in the route discovery control overhead, yielding a highly scalable operation. In turn, a limited span global route discovery process is invoked for routing shorter distance flows. Discovered global routes use effectively the capacity of non-backbone wireless links. Such an operation serves to upgrade the network’s throughput capacity level when the backbone network does not provide global topological covering. The hybrid routing protocol introduced and studied in this paper, also employs combined nodal congestion control and flow admission control schemes to guide admitted flows across areas that are less congested, and to avoid overloading the network. We present a centralized procedure as well as a distributed adaptive scheme for the calculation of the distance threshold level under varying traffic loading and backbone coverage conditions. We show our schemes to make efficient use of network-wide capacity resources by dynamically selecting proper distance threshold levels, yielding outstanding delay–throughput performance.     

A new metric for dynamic routing algorithms

A dynamic routing algorithm that has as its goal the control of congestion in a packet switching network is presented. The algorithm is based in part on the ARPANET SPF algorithm. However, instead of employing a delay metric, the authors make use of a combination of link and buffer utilizations. A detailed simulation model of the ARPANET was constructed to compare the performance of the congestion-based algorithm to the traditional delay-based (SPF) routing algorithm. The results indicate a substantial improvement in the delay and throughput of the network with the congestion-based routing algorithm     

Error and flow control performance of a high speed protocol

The performance of the SNR protocol of A. N. Netravali et al. (1990) is studied when it is implemented for end-to-end flow and error control. Using a combination of analysis and simulation, the efficiency with which this protocol uses the network bandwidth and its achievable throughput is evaluated as a function of certain network and protocol parameters. The protocol is enhanced by introducing two windows to decouple the two functions of receiver flow control and network congestion control. This enhancement and the original protocol are compared with go-back-N (GBN) and one-at-a-time-selective-repeat (OSR) retransmission procedures, are shown to have significantly higher throughput for a wide range of network conditions. As an example, for a virtual circuit with 60-ms roundtrip delay and 10^-8 bit error rate, in order to deliver 500 Mb/s throughput, both the GBN and OSR require a raw transmission bandwidth of approximately 800 Mb/s, whereas SNR with two windows needs slightly higher than 500 Mb/s raw bandwidth. Periodic exchange of state can also provide a variety of measures for congestion control in a timely and accurate fashion     

迈向PON时代的接入网

接入网连接本地交换局和用户,是用户连入电信网的第一步.通常是通过双绞线实现,双绞线对于传输话音数据是足够的,但是对于像Internet这种网络的高带宽要求就显得力不从心了.光纤骨干网接近无限的带宽,PC也以吉赫兹的速率运行,而它们之间却只通过56 kbit/s或512 kbit/s的速率连接,这就形成了接入网瓶颈.描述了如何通过光网络技术来解决接入网瓶颈问题.     

基于非强占优先消息排队系统的软交换性能分析

该文针对多进程共享处理机资源的软交换实体,提出基于非强占、多优先级消息排队的M/G/1/n排队网络性能分析模型。该模型中消息的处理服从定长分布而不是泊松分布,并且存在呼叫损失,更加接近实际系统。给出了消息平均排队时间的解析表达式,理论分析与仿真结果表明相对于无优先级M/G/1/n排队模型,上述模型具有更大系统吞吐量,更高CPU有效负荷,但呼叫接续时间稍有增加。同时分析了消息缓冲区n对于系统性能的影响。     

Acoustic burst transmission of high rate data through shallow underwater channels

Some preliminary results are presented for the transmission of data using two phase DPSK modulation at burst rates of 10 kbit/s over a short distance in shallow water. Average data rates of 1.6 kbit/s were achieved.<>     

Computer Communication Via Satellites--A Queueing Model

The last few years have witnessed the intensive growth of computer communication networks. The need for nationwide and multination computer communication systems brought about the development of packet-switching networks such as the ARPANET. In this paper we examine a model for computer-to-computer communication via a satellite link. In each network, a single node, the satellite communication concentrator (SCC), manages the flow of information between the terminals in the network and the satellite link. The SCC buffers messages from the terminals and retransmits them over the satellite channel. Buffer space claimed by a message is made free only after the SCC receives an acknowledgment from the receiving network; transmission errors cause the buffer to retransmit the message. The statistical behavior of such a system is considered.     

Directional random access scheme for mobile ad hoc networking using beamforming antennas

It has been proposed to upgrade the performance of medium access control (MAC) schemes through the use of beamforming directional antennas, to achieve better power and bandwidth utilization. In this paper, we consider a shared wireless medium as employed in a mobile ad hoc wireless network. We present and analyze a random access MAC algorithm that is combined with the use of directional beamforming formed by each transmitting mobile entity. Mathematical equations are derived to characterize the throughput performance of such a directional-ALOHA (D-ALOHA) algorithm. We describe the interferences occurring at each receiving node by considering both distance based and SINR based interference models. The D-ALOHA protocol includes the establishment of a (in-band or out-of-band) control sub-channel that is used for the transmission of location update messages. The latter is used for allowing mobile nodes to track the location of their intended destination mobiles. We present a separation property result that allows us to express the network throughput performance as a product of two factors: (1) a stationary factor that represents the system throughput performance under a perfect receiver location update process, and (2) a mobility factor that embeds the user mobility and location update processes in expressing the level of throughput degradation caused due to location update errors. We employ our derived mathematical equations, as well as carry out simulation evaluations, to present an extensive set of performance results. The throughput performance of such a beamforming based MAC protocol is characterized in terms of the system’s traffic loading conditions, the selected beamwidths of the antennas at the transmitting mobiles, the mobility levels of the nodal entities and the bandwidth capacity allocated to the control channel used for location update purposes. We show that the D-ALOHA protocol can provide a significant upgrade of network performance when the transmitting nodes adapt their beamwidth levels in accordance with our presented control scheme. The latter incorporates the involved tradeoff between the attained higher potential spatial reuse factors and the realized higher destination pointing process errors, and consequently uses nodal mobility levels and channel loading conditions as key parameters.     

Impact of relative speed on node vicinity dynamics in VANETs

Communication protocols generally rely on the existence of very long multihop paths to reach distant nodes. They disregard, however, how often such paths indeed occur, and how long they persist, especially in highly dynamic mobile networks. In this direction, this paper evaluates quantitatively the influence of node relative speed on path establishment and maintenance, using real and synthetic vehicular network traces. We propose a methodology for vehicular network analysis where both relative speeds and hop distances are used as parameters to characterize node vicinity. Results show that contact opportunities highly depend on the relative speed and the hop distance between nodes. In sparser scenarios, the number of contacts between nodes separated by more than 3 hops or even between neighbors with relative speed above 40 km/h is negligible. This confirms the intuition that contacts at lower relative speeds and at few hop distances happen more often. In addition, contacts last longer as the number of hops between nodes decreases. Nevertheless, we can still find multihop paths able to transmit messages at high relative speeds, even though less often. We also demonstrate that relative speeds reduce the number of useful contacts more severely when compared to the hop distance. For last, we show that it is possible to increase the number of successful packet transmissions by simply applying the outcomes of this work, without any sophisticated model, avoiding the waste of resources, such as energy and bandwidth.

     

Distributed Control for a Variable Length Message Store-and-Forward System

This paper describes the performance of a store-store-and-forward communications network with an adaptive routing strategy. The network chosen is a simplification of the AUTODIN network (described below) and the strategy is a modification of the routing of ARPANET. The objective is to make a network such as AUTODIN, with its wide variation in message length, effectively responsive for computercomputer and terminal-computer messages as well as record traffic. A second objective is to increase survivability following isolated equipment failures. A simulation study is used to show that improved performance is achieved.     

Subjective Evaluation of Several Efficient Speech Coders

The voice quality of several 9.6 - 32 kbit/s coders is determined with an extensive set of subjective listening tests. Single encodings of μ255 PCM, adaptive differential PCM (ADPCM), subband coding (SBC), vocoder-driven adaptive transform coding (ATC), adaptive predictive coding (APC), and time domain harmonic scaling combined with SBC are compared in an idealized situation, that is, no added impairments. It is shown that single encodings of modest complexity 32 kbit/s coders such as ADPCM and SBC and more complex 24 kbit/s coders such as vocoder-driven ATC and APC offer quality nearly equivalent to 64 kbit/s μ255 PCM. However, these conclusions are drawn in the absence of a realistic telephone network where tandem encodings, delay limitations, and nonvoice signals exist. Tandem encodings of 64 kbit/s μ255 PCM, 32 kbit/s ADPCM, 16 kbit/s SBC, and 16 kbit/s APC are also evaluated. These 32 kbit/s and 16 kbit/s coders offer degraded tandem performance as compared to 64 kbit/s PCM, with the exception of synchronous tandeming of 32 kbit/s ADPCM with 64 kbit/s PCM where several encodings are subjectively equivalent to a single encoding of 32 kbit/s ADPCM.     

Analysis and Simulation of a Plastic Optical Fiber Access Network Based on OCDMA Technology

In this paper a plastic optical fiber (POF) access network based on optical code-division multiple access (OCDMA) technology is proposed. Recently, optical transmission using POF has received much attention due to its low weight, large core diameter, flexibility, and easy installation. Specially, its high bandwidth makes POF a very attractive candidate for a transmission media in an access network based on OCDMA technology. A conventional OCDMA system only allows a finite number of units to transmit and access simultaneously according to the number of channels which is restricted by BER. To resolve this problem a novel protocol is also proposed in this paper. The protocol can efficiently support variable-sized messages, and any new unit can join the network at any time without requiring network initialization. To implement the demonstration, each optical network unit is equipped with a fixed and a tunable optical encoder/decoder. The optical encoder/decoder employing planar holographic optical processors (HOPs) in this system may be of low fabrication cost. The network throughput and average delay using various system parameters has been investigated by numerical analysis and simulation experiments. It is shown that the dynamic control protocol in this POF access network based on OCDMA technology is valid and efficient.     

Supporting real-time and multimedia applications on the Mercuritestbed

This paper describes the distributed system, network and software architecture, the application development environment, the performance, and the early lessons learned on the ATM LAN testbed Mercuri established at the Honeywell Technology Center, to develop distributed multimedia technologies for real-time control applications. We have developed a client-server-based software architecture on Sun Sparcstation-2s connected by a Fore Systems' ASX-100 ATM switch, with video processing handled by Parallax's X Video cards. The architecture enables network-transparent applications and provides simple primitives for multimedia capture, display, transmission, storage, and retrieval. A real-time multimedia-in-the-loop control application was developed as the vehicle for testing the capabilities and performance of the network. Our test measurements focus on the end-user-level performance metrics such as message throughput and round-trip delay as well as video-frame jitter under no-load and load conditions. Our results show that the maximum burst throughput that can be supported at the user level is 48 Mb/s using AAL 5, while round-trip delays for 4-kbyte messages are about 3 ms. Our experience reveals a number of performance bottlenecks and open issues in using commercial ATM switches for practical applications. Our conclusions are outlined     

DRP: An energy‐efficient routing protocol for underwater sensor networks

The features of transmissions in underwater sensor networks (UWSNs) include lower transmission rate, longer delay time, and higher power consumption when compared with terrestrial radio transmissions. The negative effects of transmission collisions deteriorate in such environments. Existing UWSN routing protocols do not consider the transmission collision probability differences resulting from different transmission distances. In this paper, we show that collision probability plays an important role in route selection and propose an energy‐efficient routing protocol (DRP), which considers the distance‐varied collision probability as well as each node's residual energy. Considering these 2 issues, DRP can find a path with high successful transmission rate and high‐residual energy. In fact, DRP can find the path producing the longest network lifetime, which we have confirmed through theoretical analysis. To the best of our knowledge, DRP is the first UWSN routing protocol that uses transmission collision probability as a factor in route selection. Simulation results verify that DRP extends network lifetime, increases network throughput, and reduces end‐to‐end delay when compared with solutions without considering distance‐varied collision probability or residual energy.     

Streaming Multimedia for the Internet

Multimedia streaming enables Web sites to be enhanced with real-time audio and video content. Through the appropriate use of video compression technologies, audio and video content can be delivered directly into the Web browser environment to create a rich mix of text, graphics and multimedia. The audio quality and video window size available to viewers is primarily determined by network access speeds. For the Internet, audio and video can be successfully streamed at rates between 20 kbit/s and 37 kbit/s over a PSTN modem link. In the corporate intranet environment, where higher end-to-end network speeds exist, audio and video can easily be streamed at rates between 56 kbit/s and 450 kbit/s. This paper illustrates how commodity video compression techniques facilitate the introduction of real-time multimedia streaming over IP networks. It describes the technology components within a basic video streaming system and provides an overview of two applications that utilise this technology, namely Intranet TV and BT Results Webcast. The underlying network protocols required to support real-time Internet applications are also discussed and network measurement statistics obtained from Intranet TV and BT Results Webcast are also presented.     

Transmission quality of north American cellular, personalcommunications, and public switched telephone networks

Future long distance, and especially international calls, will involve an increasing number of multilink circuits of cellular, personal communications, mobile satellite, and public switched telephone network (PSTN) type of connections incorporating a variety of speech coding devices. In particular, the rapid growth of cellular communications has highlighted the need to characterize the quality of switched networks when cellular terminals are attached at their termination nodes. At the same time, the nonlinear nature of low-rate parametric speech coding has rendered questionable analytical methods for estimating end-to-end voice quality of interconnected telecommunications networks. Instead, quantification of transmission performance appears to require direct subjective evaluation of the pertinent conditions of interest. In this paper the quality of interconnected North American digital cellular and future microcellular terminals with 16 kbit/s and 32 kbit/s DCME/PCME-based switched and private telephone networks is quantified. From these assessments it can be concluded that cellular networks employing the TIA IS-54 8 kbits/s VSELP algorithm may meet the end-to-end transmission planning criteria when interconnected with the switched network     

The Economics and Performance of Satellite Packet Switching

The economics and performance of packet-switched satellite systems operating in a broadcast mode are studied and compared to landline-based systems similar to the ARPANET. The fixed-packet length inefficiencies and the satellite channel utilization of the satellite systems are presented as functions of the message length distribution, the number of earth stations, the average message delay, and the average number of messages generated at each earth station. Some of the economic factors in the design of packetswitched satellite networks are discussed, including hardware costs as a function of technology and system performance. Satellite network configurations with or without backhaul facilities are compared and it is concluded that configurations using backhaul facilities consisting of two 30 mile private leased lines (each 50 kbits/s) are more expensive. Parameters and estimated costs of satellite system configurations are presented and an engineering estimate of the cost function for satellite packet-switched networks is derived and presented in the form of long range average cost curves. Economies of scale with respect to network traffic and diseconomies of scale with respect to the number of network earth stations are found to be present.     

An Approach to Multi-Service Subscriber Loop System Using Packetized Voice/Data Terminals

A new subscriber communication system and its design philosophy are described. In addition to telephone communication, the system is simultaneously able to offer data communication, still picture communication, etc. The system is composed of a packetized voice/data terminal, a multi-service switching equipment and the digital subscriber loop connecting between them. The system utilizes the existing subscriber line efficiently and is entirely suitable for coming telephone and data public digital network. 64 kbit/s PCM coded voice conversation and 48 kbit/s data communication were achieved simultaneously over 96 kbit/s digital subscriber line which was spanned up to 2 km.