In this paper, we propose a model of how bio-nanomachines differentiate their functionality in molecular communication. In the proposed model, bio-nanomachines perform a set of basic functions such as releasing signal molecules into the environment, detecting the concentration of signal molecules in the environment, and migrating toward the higher concentration of signal molecules, while they acquire energy from the environment and consume energy to perform such functions. The net amount of energy (total energy acquired minus total energy consumed) of a bio-nanomachine reflects what functions it performed and how actively it performed such functions and indicates how it will differentiate itself from other bio-nanomachines in performing functions. Simulation results show that bio-nanomachines starting with the same amount of net energy come to have different amounts of net energy and, as a result, differentiate and form a complex spatial distribution of bio-nanomachines. The proposed model helps understand how bio-nanomachines differentiate and acquire complex functionalities in molecular communication.

In this paper, we consider kinesin molecular motors transporting a bead over a network of microtubules and develop mathematical models to describe the stochastic behavior of kinesins carrying a bead. In developing the mathematical models, we consider the kinesins' stochastic behavior (1) at an intersection of crossing microtubules to obtain the probabilities of pass, switch, pause and dissociate stochastic behaviors of the kinesins carrying a bead and (2) over a network of microtubules through multiple microtubule intersections to obtain transition probabilities for kinesins carrying a bead starting from a given sender to reach a given receiver in the network. We examine the accuracy of the developed mathematical models using in vitro experimental results in our previous work.

This paper explores the dynamic properties of oscillating and propagating patterns in concentration of the information molecule and proposes a new modulation scheme for molecular communication. In the proposed modulation scheme, information molecules transmitted from a sender bio-nanomachine propagate in the environment, chemically react with the molecules in the environment, change their concentration, and form an oscillating and propagating pattern in the environment. A sender bio-nanomachine modulates information onto the dynamic properties of the pattern, such as the amount of change in concentration of the information molecule and the time duration of one cycle of a periodic pattern in concentration of the information molecule. Receiver bio-nanomachines receive the oscillating and propagating pattern that the sender bio-nanomachine generates and detect its dynamic properties to demodulate information. This paper implements the proposed modulation scheme using Ca2+oscillation and propagation, a communication mechanism found in biological cells. It also demonstrates that the proposed modulation scheme allows information to be simultaneously modulated onto multiple dynamic properties of an oscillating and propagating pattern that a sender bio-nanomachine generates. The main contributions of this paper are to open a new research area of modulation schemes in molecular communication and to illustrate the possibility that biological cells use such modulation schemes for cell-to-cell communication.

In IaaS cloud services, QoS of network applications (NW-Apps) may degrade due to location factors such as significant distance between a server-side application (server) of a NW-App at a data center and a client-side application (client) of the NW-App at a client terminal. In order to shorten the distance and to improve the QoS, server migration services (SMSes) have been proposed. In SMSes, servers may migrate between different computers (called work places, WPs) on a network to prevent QoS degradation caused by the changes of client locations. Although server migrations can improve QoS of NW-Apps, they also generate a huge amount of traffic (server migration traffic) in the network. This paper focuses on a server location decision problem where the location of a server is decided in an on-line manner so that QoS of a NW-App is improved under the constraint that the server migration traffic has to be suppressed below an acceptable level. For the problem, we propose a practical on-line algorithm. The key idea behind the proposed algorithm is that the location of the server is decided with consideration of the QoS degradation in the future. The algorithm defines the averagely good location for the server where the QoS is expected to be relatively good for various client locations. Then, it keeps the range of the server's migration within the returnable range where the server can soon come back to the averagely good location. As a result, the QoS can be always kept as good as the one under the averagely good location. Simulation results show that the proposed algorithm improves QoS of the NW-App by up to 30% compared to a greedy algorithm.

This paper proposes to use spatio-temporal patterns that the concentration of propagating information molecules form in the molecular communication environment and develops a new modulation technique for molecular communication between bio-nanomachines. In molecular communication considered in this paper, information molecules transmitted from a group of sender bio- nanomachines propagate in the environment, chemically react with the molecules in the environment, change their concentration, and form an oscillating and propagating pattern. The rates at which sender bio-nanomachines transmit information molecules determines the frequency, amplitude and phase characteristics of the pattern, and the sender bio-nanomachines modify the transmission rates in order to modulate information on to frequency, amplitude and phase characteristics of the pattern. A group of receiver bio-nanomachines detects these characteristics to collectively decode information. This paper develops a new model of molecular communication through pattern formation. Using the model developed in this paper and numerical examples, this paper demonstrates the advantages of the proposed modulation technique, namely, higher information capacity and longer communication distances.

In molecular communication, a group of biological nanomachines communicates through exchanging molecules and collectively performs application dependent tasks. An open research issue in molecular communication is to establish interfaces to interconnect the molecular communication environment (e.g., inside the human body) and its external environment (e.g., outside the human body). Such interfaces allow conventional devices in the external environment to control the location and timing of molecular communication processes in the molecular communication environment and expand the capability of molecular communication. In this paper, we first describe an architecture of externally controllable molecular communication and introduce two types of interfaces for biological nanomachines; bio-nanomachine to bio-nanomachine interfaces (BNIs) for bio-nanomachines to interact with other biological nanomachines in the molecular communication environment, and inmessaging and outmessaging interfaces (IMIs and OMIs) for bio-nanomachines to interact with devices in the external environment. We then describe a proof-of- concept design and wet laboratory implementation of the IMI and OMI, using biological cells. We further demonstrate, through mathematical modeling and numerical experiments, how an architecture of externally controllable molecular communication with BNIs and IMIs/OMIs may apply to pattern formation, a promising nanomedical application of molecular communication.

This paper investigates, through computer simulations, statistical behavior of molecular motors transporting cargos on a network of protein filaments in an engineered environment. This paper also examines how molecular motors' statistical behavior is applied to building. a molecular communication system. A molecular motor, such as kinesin, is a protein complex. It exists in eukaryotic cells and functions as a nano-scale autonomous nanomachine that carries nano-scale to micro-scale signaling molecules on a network of protein filaments (i.e., cytoskeletal network) in eukaryotic cells. At an intersection of protein filaments on a cytoskeletal network, a molecular motor shows stochastic movement and may stay on the current protein filament or may switch onto an intersecting protein filament, resulting in stochastic delivery of signaling molecules from the nucleus to organelles over a cytoskeletal network. Molecular communication is an engineered system to transmit information by modulating information on molecules and by transporting modulated molecules from senders to receivers. One of promising approaches to building a molecular communication system is to use molecular motors to transport information-modulated molecules on an engineered protein filament network connecting senders and receivers. In order to examine the feasibility of this approach, this paper simulated statistical behavior of molecular motors carrying cargo molecules through multiple protein filament intersections on a protein filament network. Simulations model stochastic movement of a molecular motor at a single intersection of protein filaments using the measurement data obtained through in vitro experiments (using kinesins and microtubules) conducted by the authors of this paper and evaluates the stochastic delivery of molecules from the senders to the receivers over multiple intersections of protein filaments arranged in an array topology. Simulation results show that molecular motors stochastically deliver cargo molecules to a specific receiver (or to a specific set of receivers) with a given probability over an array of protein filament intersections.

Technology is taking us to a world where myriads of heavily networked devices interact with the physical world in multiple ways, and at multiple scales, from the global Internet scale down to micro- and nano-devices. Many of these devices are highly mobile and autonomous, and must adapt to the surrounding environment in a totally distributed and unsupervised way. fundamental research challenge is the design of robust decentralized computing systems, which are capable of operating under changing environments, and yet exhibit the desired behavior and response time, under unpredictable operating constraints, such as traffic demand, energy consumption, size, and processing power. These systems should be able to adapt and learn how to react to unforeseen scenarios, as well as to display desired emergent properties.

There is currently much debate in defining what form the future Internet will take. The current Internet is struggling to meet the needs of an ever-evolving society. This is largely due to the Internet now become a thriving marketplace with services at the core. The range, number and complexity of services are set to increase with an even more dynamic service environment envisioned in the future. However, as these services grow, service composition will become an important feature of the service environment, leading to new challenges in service discovery and composition mechanisms. At the same time, dynamic service environments will also require that the underlying infrastructure networks are flexible enough to handle the changing service landscape. One area this is particularly important is in dynamic routing to deal with highly dynamic and frequent service changes. In this paper, we adopt mechanisms from biology and apply these to the problems identified, resulting in an integrated bio-inspired service management and dynamic routing solution for Future Internet. We demonstrate how the bio-inspired mechanisms not only improve each problem individually, but through their integration also improve overall network performance. Simulation results are presented to validate the proposed solution.

In recent years, the notion of service overlay networks has been proposed as a promising solution for providing end-to-end QoS without changing the current Internet architecture. A major issue in deploying service overlay networks is determining how to allocate resources (such as link bandwidth) on a substrate network to overlay networks, while satisfying the end-to-end QoS requirements of applications running on each overlay network. This paper introduces the Market-based Cooperative Resource Allocation (MaCRA) architecture that achieves fair and efficient resource allocation in a decentralized manner. In MaCRA, resources on a substrate network are priced, and each overlay network provider creates an overlay network on a minimum cost basis to meet its application QoS requirements. MaCRA also allows each overlay network provider to trade their current resources with other overlay network providers when resources on a substrate network are not available or expensive. Simulation results demonstrate that MaCRA achieves fairness and efficiency in allocating resources for overlay networks when compared to existing mechanisms.

Storing data on sensor nodes at a specified location is a useful technique for data-centric storage and management of location dependent data on wireless sensor networks. To decrease the cost of accessing these data, arranging replicas of data so that the distance between the consumer nodes and replica holder nodes decreases is desirable. Conventional schemes, however, are costly for updating the replicas even if they are not often used or do not support functions for ensuring their consistency. We propose a novel replica arrangement scheme, which adaptively arranges replicas at positions close to nodes for frequent sending of queries. Instead of arranging many replicas on sensor nodes, our scheme consists of nodes with pointers that point to the replica holder nodes in order to save storage for replicas. The pointers, which are smaller than replicas, are arranged so that nodes are in circular arcs surrounding the location where the original data item is associated. Simulation results show that our scheme outperforms conventional replica arrangement schemes in terms of the cost required for sending queries and replies with sufficiently low replica update cost.

This article presents a semantics-based context-aware dynamic service composition framework that composes an application through combining distributed components based on the semantics of components and contexts of users. The proposed framework consists of Component Service Model with Semantics (CoSMoS), Component Runtime Environment (CoRE), and Semantic Graph based Service Composition (SeGSeC). CoSMoS models the semantics of components and contexts of users. CoRE is a middleware to support CoSMoS on various distributed computing technologies. SeGSeC is a mechanism to compose an application by synthesizing its workflow based on the semantics of components and contexts of users. The proposed framework is capable of composing applications requested in a natural language by leveraging the semantic information of components. The proposed framework composes applications differently to individual users based on their contexts and preferences. The proposed framework acquires user preferences from user-specified rules and also via learning. The proposed framework also adapts to dynamic environments by autonomously composing a new application upon detecting context change. This article describes the design and mechanism of the proposed framework, and also presents simulation experiments to evaluate the proposed framework.

The inter-domain Resource Exchange (iREX) architecture uses economic market mechanisms to automate the deployment of end-to-end (E2E) inter-domain (ID) quality of service (QoS) policy among resource consumer and resource provider Internet Service Providers (ISPs). In iREX, each policy reservation is deployed on a single E2E ID path made up of the most "desirable" (i.e. cheapest and least congested) ISP resources. To accommodate ISPs that prefer redundancy when deploying ID QoS policy, in this paper we introduce an extension to the iREX architecture that gives an originating ISP a multi- path option (MPO) when deploying a reservation. MPO takes an initiating ISP's preference for redundancy and provides information about the available path options to achieve this preference in a distributed manner. Our simulation results show that while providing redundancy to the originating ISP using MPO does increase its resource costs in accordance to an ISP's preference, it only marginally increases overhead, and does not affect overall network performance - in fact the use of MPO lowers congestion.

Molecular communication is one method for communication among nano-scale components (artificial or biological) that perform simple computation, sensing, or actuation. Future applications using nano-scale components may require communication mechanisms to coordinate with other components. For example, uni-cast is one primitive communication mechanism for transmission from one sender to one receiver and allows two nano-scale components to coordinate and perform some function. In this paper, we consider designing a molecular communication system that applies a molecular motor transport mechanism existing in biological cells. In molecular motor transport, a sender (nano-scale component) releases information molecules, and molecular motors transport the information molecules along protein filaments to a receiver (nano-scale component) up to hundreds of micrometers away. We consider a plus-centered aster (receiver with multiple protein filaments leading to the receiver) as one possible arrangement for protein filaments. We perform simulations to evaluate the probability of sending an information molecule to the receiver. The simulation results indicate that the proposed uni-cast molecular communication is limited in range (distance between sender and receiver with a reasonable success probability) and that the proposed molecular motor system transports simulated information molecules (100 nm radius spheres) with a greater success probability and than a passive diffusion-only system.

Internet protocol-based multimedia applications are gaining momentum thanks to the great proliferation of the Internet and mass deployment of broadband. The abundance of rich content, heterogeneity of networks, user terminals, and services, along with the surges of multimedia traffic, have convoluted ever-increasing complexity and costs in the operation and management of multimedia systems. Autonomic communications (AutoComm) promotes the autonomy of communication networks with minimum human administration. In a multimedia system that features AutoComm, the networks and their elements strive to realize efficiency, immunity, resilience, and evolvability through intrinsic self-governance. By encapsulating complexity within the networks, exploiting their intelligence, and using technology to manage technology, human administrative efforts can be greatly reduced while management efficiency can be maximized. In this paper, we want to illustrate how traditional way of management and control of multimedia networks can be gradually replaced by network autonomy and self-management. Related work is extensively reviewed and compared to our approach. Initial experiment results are presented as proof of concept.

The inter-domain resource exchange (iREX) architecture uses economic market mechanisms to automate the ad-hoc negotiation and deployment of end to end inter-domain quality of service policy among resource consumer and resource provider . In this paper, we explore iREX's network load distribution by comparing its performance to a lower bound for network congestion in two ways. We first present an analytical model of iREX in terms of an online algorithm and analyze its efficiency via competitive analysis. Our main result shows that the efficiency loss of iREX with respect to monetary cost is upper-bounded by a factor of 8 K/2 K+1, where K s the number of deployments, provided affine linear price functions are used. When the price functions are used to model congestion in the network, this result implies upper bounds on the efficiency loss of iREX with respect to network congestion. We then complement the analytical model with a numerical study using simulations.with optimal solutions derived from unsplittable and splittable multi-commodity flow optimization models. Our numerical results show that for nominal to high traffic loads of 40% or more, iREX deviates a maximum of about 20% from the lower bound, while the current method deviates a maximum of 300%.

The utility of a sensor network derives primarily from the sensing data it gathers. Therefore, a critical issue of sensor networks is to efficiently provide sensing coverage. This paper proposes a scheme, enhanced differentiated surveillance (EDS), to maintain the required coverage for sensor networks by establishing working schedules of sensors for the purpose of energy saving. EDS allows every sensor in a sensor network to establish its working schedule in a distributive manner, based on random reference times generated through integer hashing and a proposed coverage measurement rule. The required coverage is continuously guaranteed when every sensor senses following its own working schedule.While most existing coverage maintenance schemes are only applicable for static sensors with equal sensing range (homogeneous sensors), EDS is versatile in applications. It can be applied to sensors with different sensing ranges (heterogeneous sensors), and extended to random mobile sensor networks, where sensors randomly roam and cannot control their movements. Compared with existing work, the advantages of EDS lie in minimum communication overhead, fast convergence, load balancing, and high battery efficiency. The effectiveness and efficiency of EDS are investigated through numerical examples of static and random mobile sensor networks.

The utility of a wireless sensor network derives primarily from the sensing data it gathers. Therefore, a critical issue for wireless sensor networks is to efficiently provide sensing coverage. This paper proposes a scheme, enhanced differentiated surveillance (EDS), to maintain the required sensing coverage by establishing working schedules of sensors for the purpose of energy saving. In EDS, each sensor generates its working schedule in a distributive manner with only one message exchanged among neighboring sensors. The required coverage is continuously guaranteed when every sensor schedules on/off sensing states according to its working schedule. Compared to the existing coverage maintenance schemes, EDS exhibits the beneficial features of minimum communication overhead, fast convergence, load balancing, and an efficient battery discharge pattern. Integrated with a neighbor discovery protocol, EDS is applied to random mobile sensor networks, where sensors randomly roam and cannot control their movements. While there has been much research effort on coverage maintenance by utilizing mobile sensors that can automatically move, coverage maintenance in random mobile sensor networks has not been addressed in the literature. Simulation results establish EDS as an effective coverage maintenance scheme for static and random mobile sensor networks.

In the current Internet, congestion control is performed jointly by the end systems running the TCP protocol and by routers running active queue management (AQM) algorithms. Due to the TCP protocol's AIMD congestion control algorithm and its round trip time delay to react to packet losses, it is very hard to maintain high TCP throughput with a low packet loss rate in routers. In addition, unresponsive traffic, such as short HTTP sessions, do not react to AQM packet loss/marks. Hence, these unresponsive traffic may cause high packet loss in the AQM queues due to its high bursts. In this paper, we provide a solution to identify unresponsive traffic in AQM queues that does not need packet header examination. In our solution, periodic features of TCP traffic due to its AIMD behavior is first estimated in the mixed incoming traffic. We then design a wavelet de-noising filter to separate the high bursts in unresponsive traffic from the TCP traffic and allow those bursts to bypass the AQM queue. By allowing bursts of unresponsive traffic to bypass AQM queues, we not only avoid the impact of unresponsive traffic to AQM queues, but also avoid dropping extra packets in those unresponsive flows. Our proposed de-noising scheme is suitable for high speed networks, where per packet header examination is expensive. Our simulations show that the proposed de-noising scheme is effective on heavily congested links. An analysis of the computational complexity of the proposed scheme is also provided.

Molecular communication is a new paradigm for communication between biological nanomachines over a short-range (a nano- and micro-scale range). Biological nanomachines are nano- and micro-scale devices that either exist in the biological world or are artificially created from biological materials and that perform simple functions such as sensing, logic, and actuation. Molecular communication provides a mechanism for biological nanomachines to communicate information by propagating molecules that represent the information. Molecular communication is based on observations of existing biological systems which use molecules as communication carriers. With the advancement of current research in areas such as synthetic biology and bio-nanotechnologies, it may become relatively easy in the near future to develop systems of biological nanomachines communicating through molecules. In this paper, we present a framework for describing molecular communication systems.

Energy efficiency is one of the most challenging issues in wireless sensor network as the sensors have to serve unattended. Cluster based communication can reduce the traffic on the network and gives the opportunity to other sensors for periodic sleep and awake and thus saves energy. Passive clustering is less computational and light weight. In cluster based approach, cluster heads and gateways have to have maximum energy to be awaken for all the times. Existing passive clustering algorithm uses first declaration method without any priority generates severe collisions in the network and form the clusters very dense with large amount of overlapping regions. This results increased number of gateways. We have proposed several modifications for the existing passive clustering algorithm to prolong the life time of the network with better cluster formation. More -over, our proposed solution finds the optimum path between sources and sinks using the tiny cache memory of the intermediate nodes. Simulation result shows that EAREC saves significant amount of energy and at the same time keeps the delay and success rate satisfactory.

This paper introduces the inter-domain Resource Exchange (iREX) architecture for the automated deployment of end to end (E2E) inter-domain (ID) quality of service (QoS) policy among multiple Internet Service Providers (ISPs). iREX uses economics and fully distributed mechanisms to allow source domains to self-manage the deployment of E2E ID QoS policy while promoting congestion-avoidance by employing a distributed resource selection process that selects the least congested ID resources for deployment paths. With iREX, domains advertise its available ID QoS network resources for sale to their neighboring domains. Domains receiving these advertisements evaluate and filter the advertised resources according to their price and reputation before disseminating the advertisements to neighboring domains. Source domains needing to deploy E2E ID QoS policy use the advertised resource information to select a source routed path to the destination and establish bilateral agreements directly with each resource provider along the path.

The inter-domain resource exchange (iREX) architecture uses economic market mechanisms to automate the deployment of end to end (E2E) inter-domain (ID) quality of service (QoS) policy among resource consumer and resource provider Internet Service Providers (ISPs). Previous simulation results have shown that iREX allows more coexisting ID policy deployments with less network congestion when compared to the existing method. In this paper we explore iREX's network load distribution efficiency limits by comparing iREX's performance to a lower bound for network congestion. We present an analytical model of iREX in terms of a min-cost flow problem, and numerical results of efficiency loss between iREX simulations and derived optimal solutions based on multi-commodity flow optimization models. Our results show that for nominal to high traffic loads of 50% or more, iREX deviates a maximum of approximately 30% from the derived lower bound, while the current method deviates a maximum of 350%.

Owing to numerous potential applications, wireless sensor networks have been attracting much research effort recently. The critical challenge that wireless sensor networks often face is to sustain long-term operation on limited battery energy. Coverage maintenance schemes can effectively prolong network lifetime by maintaining sufficient sensing coverage over a target region using a small number of active sensors while scheduling the other sensors to sleep. We envision future wireless sensor networks are composed of a vast number of extremely miniaturized sensors (e.g., millimeter-scale) deployed in exceedingly high density (e.g., more than 1000sensors/m2). Therefore, the key issue of coverage maintenance for the future sensor networks is the scalability to sensor deployment density. In this paper, we propose a novel coverage maintenance scheme, called scalable coverage maintenance (SCOM). SCOM is energy efficient and scalable to sensor deployment density in terms of communication overhead (i.e., number of transmitted and received beacons) and computational complexity (i.e., time and space complexity). We validate our claims through both analysis and simulations

This paper presents a semantics-based dynamic service composition architecture that composes an application through combining distributed components based on the semantics of the components. This architecture consists of a component model called Component Service Model with Semantics (CoSMoS), a middleware called Component Runtime Environment (CoRE), and a service composition mechanism called Semantic Graph based Service Composition (SeGSeC). CoSMoS represents the semantics of components. CoRE provides interfaces to discover and access components modeled by CoSMoS. SeGSeC composes an application by discovering components through CoRE, and synthesizing a workflow of the application based on the semantics of the components modeled by CoSMoS. This paper describes the latest design of the semantics-based dynamic service composition architecture, and also illustrates the implementation of the architecture based on the Web Service standards, i.e. WSDL, RDF, SOAP, and UDDI. The Web Service based implementation of the architecture allows existing Web Services to migrate onto the architecture without reimplementation. It also simplifies the development and deployment of a new Web Service on the architecture by automatically generating the necessary description files (i.e. WSDL and RDF files) of the Web Service from its runtime binary (i.e. a Java class file).

Distributed collaborative applications will be widely used in a future large-scale and dynamic network. Such applications require the capability of locating distributed network objects that better match query originator's criteria. In this paper, a discovery mechanism utilizing query originator's preference is proposed in order to locate objects preferred by query originators. The proposed mechanism allows query originators to return feedbacks that describe the degree of the preference for discovered objects. The preference of query originators is utilized to guide subsequent queries in a distributed manner, enhancing efficiency of discovery. The preference of query originators is also utilized to adaptively change forwarding paths of queries corresponding to varying network environments. Extensive simulation work shows that the proposed mechanism is efficient and adaptive to dynamics of the peer-to-peer network environment where transient objects often join and leave.

Multi-rate feature is an important advance in wireless communications. It enables wireless devices to operate at high data rate when the channel condition is sufficiently good. Existing ad hoc routing protocols can not perform very well in the multi-rate ad hoc networks because they generally assume that bandwidth of different links are identical in the system. One straightforward solution to this issue is to use special routing metrics in routing algorithm, which is called routing approach. However, we believe that only localized reaction is performed in response to the local event. Based on this design principle, we propose a relay based medium access scheme to employ multi-rate capability even if existing minimum-hop based routing protocols are used. This scheme takes advantage of the node residing between neighboring nodes as a relay node and transforms a long-range transmission into two short-range transmissions. In our scheme design, we are very careful in reducing the additional overhead caused in order to exploit the high data rate transmission. Simulations show that our scheme achieve better throughput compared with 802.11 MAC scheme, a rate-adaptive MAC protocol called RBAR and the routing approach.

Finite battery energy and limited bandwidth resources are the two major constraints in ad hoc networks. Therefore, researchers have persistently sought for optimization algorithms to reduce the control overhead and to increase bandwidth utilization efficiency, using mechanisms such as topology control and management, multi-rate adaptation etc. to cut down the energy expenditure in ad hoc networks. However, the energy conservation and network throughput improvement are handled separately in many cases. In this paper, we propose a unified framework that combines the multi-rate adaptation and clustering mechanisms so as to provide the optimal network throughput under very low control overhead. The framework incorporates two control mechanisms into the networking stack. The first mechanism is based on a novel clustering algorithm, called PATM (priority-based adaptive topology management), that constructs the backbone of the network topology for routing protocols. The second mechanism is a medium access control protocol, called RMAC (relay-based MAC), that provides high data transmission rates between adjacent backbone nodes. The efficiency of the unified framework is evaluated using theoretical analysis and extensive simulations with DSR (dynamic source routing) as the underlying routing protocol. We demonstrate that our solution not only significantly reduces the routing control overhead, but also substantially improves the network throughput.

Complex services may be dynamically composed through combining distributed components on demand (i.e., when requested by a user) in order to provide new services without preinstallation. Several systems have been proposed to dynamically compose services. However, they require users to request services in a manner that is not intuitive to the users. In order to allow a user to request a service in an intuitive form (e.g., using a natural language), this paper proposes a semantics-based service composition architecture. The proposed architecture obtains the semantics of the service requested in an intuitive form, and dynamically composes the requested service based on the semantics of the service. To compose a service based on its semantics, the proposed architecture supports semantic representation of components [through a component model named Component Service Model with Semantics (CoSMoS)], discovers components required to compose a service [through a middleware named Component Runtime Environment (CoRE)], and composes the requested service based on its semantics and the semantics of the discovered components [through a service composition mechanism named Semantic Graph-Based Service Composition (SeGSeC)]. This paper presents the design, implementation and empirical evaluation of the proposed architecture.

This work describes and empirically evaluates the middleware platform of a new network architecture called the Bio-Networking Architecture. The Bio-Networking Architecture is inspired by the observation that the biological systems (e.g., bee colonies) have already developed mechanisms necessary to achieve future network requirements such as autonomy, scalability, adaptability, and simplicity. In the Bio-Networking Architecture, a network application is implemented as a group of distributed, autonomous and diverse objects called cyber-entities (CEs) (analogous to a bee colony consisting of multiple bees). Each CE implements a functional service related to the application and follows simple behaviors similar to biological entities (e.g., reproduction and migration). In the Bio-Networking Architecture, beneficial application characteristics (e.g., autonomy, scalability, adaptability, and simplicity) arise from the autonomous interaction of CEs. The middleware platform in the Bio-Networking Architecture, the bionet platform, provides reusable software components for developing, deploying, and executing CEs. The components abstract low-level operating and networking details, and implement high-level runtime services that CEs use to perform their services and behaviors. The components in the bionet platform are designed based on several biological concepts (e.g., energy exchange and pheromone emission). This work describes key designs of the bionet platform and empirically demonstrates that the bionet platform is efficient, scalable, reusable, and significantly simplifies development of network applications.

This paper proposes a novel framework for developing adaptive and scalable network services. In the proposed framework, a network service is implemented as a group of autonomous agents that interact in the network environment. Agents in the proposed framework are autonomous and capable of simple behaviors (e.g., replication, migration, and death). In this paper, an evolutionary adaptation mechanism is designed using genetic algorithms (GAs) for agents to evolve their behaviors and improve their fitness values (e.g., response time to a service request) to the environment. The proposed framework is evaluated through simulations, and the simulation results demonstrate the ability of autonomous agents to adapt to the network environment. The proposed framework may be suitable for disseminating network services in dynamic and large-scale networks where a large number of data and services need to be replicated, moved, and deleted in a decentralized manner.

In reactive routing protocols for ad hoc networks, the network wide flooding generates a significant amount of traffic overhead. The goal of This work is to reduce this overhead. We propose a path based route discovery scheme which can find a route to the destination with lower overhead than flooding-based route discovery. We also showed that the probability of finding a path to the destination by using path based routing request is not negligible. The path obtained through path based routing request may be far from optimal. Thus, we propose a route expanding scheme which can discover a better path based on the path discovered in path based routing discovery phase. By simulation, we showed that integrating these two schemes has better performance than existing reactive routing protocols like DSR.

Dynamic composition of complex services from primitive components brings flexibility and adaptability to future applications. By properly selecting and combining components on dand, applications would adapt to individual user preference and would consider available context information. Existing service composition systs often require users to request services in strict syntax formats, such as data types, service tplates or logic formulas. This requirent may become an obstacle for end-users to use such systs. Instead, service composition should be santics-based so that a service is requested and composed not by its syntax but by its santics. 0In order to enable santics-based dynamic service composition, both the modeling of components as well as the service composition mechanism must support santics. To satisfy the requirent of santic support in the component modeling, we have designed a new model named Component Service Model with Santics (CoSMoS). CoSMoS integrates the santic information of a component and the functional information of a component into a single santic graph representation. A unified interface named Component Runtime Environment (CoRE) is developed to convert different component implentations onto the CoSMoS representation. Using the santic support of CoSMoS, we have developed a santics-based service composition mechanism named Santic Graph based Service Composition (SeGSeC). SeGSeC generates the execution path of the requested service, and checks the santics of the path against the request. We have implented a service composition syst using the above techniques, and donstrated that our syst supports santics-based dynamic service composition.

This paper proposes a new QoS framework, called the on-demand QoS path framework (ODP). ODP provides end-to-end QoS guarantees to individual flows with minimal overhead, while keeping the scalability characteristic of Diff-Serv. ODP exercises per-flow admission control and end-to-end bandwidth reservation at the edge of the network and only differentiates traffic classes in the core of the network. In addition, to adapt to dynamically changing traffic load, ODP monitors the bandwidth utilization of the network and performs dynamic bandwidth reconfiguration in the network core. Through extensive simulations, the performance of ODP is investigated and compared with that of IntServ and DiffServ frameworks. The simulation results clearly show that ODP provides end-to-end QoS guarantees to individual flows, which DiffServ can not provide, with much less overhead than IntServ.

Supporting transmission of stream media over wireless mobile networks is often difficult because packets may be lost due to the rerouting of packets during handoff, and also because bursts of packet loss may occur during handoff due to the disparity in the amount of available bandwidth among different cells. In this paper, we propose an end-to-end multipath handoff scheme that provides smooth handoff for stream media in wireless networks with different amounts of available bandwidth from cell to cell. In the proposed scheme, multiple paths are established during handoff to reach a mobile destination node. The stream media sources are equipped with an adaptive multilayer encoder, and important layers in the encoded video stream are duplicated and transmitted over multiple paths during handoff. The effectiveness of the proposed multipath handoff scheme is verified and compared with existing schemes through extensive simulations. The simulation results show that the proposed scheme provides higher throughput and better quality for stream media.

The Internet's excellent scalability and robustness result in part from the end-to-end nature of Internet congestion control. End-to-end congestion control algorithms alone, however, are unable to prevent the congestion collapse and unfairness created by applications that are unresponsive to network congestion. To address these maladies, we propose and investigate a novel congestion-avoidance mechanism called network border patrol (NBP). NBP entails the exchange of feedback between routers at the borders of a network in order to detect and restrict unresponsive traffic flows before they enter the network, thereby preventing congestion within the network. Moreover, NBP is complemented with the proposed enhanced core-stateless fair queueing (ECSFQ) mechanism, which provides fair bandwidth allocations to competing flows. Both NBP and ECSFQ are compliant with the Internet philosophy of pushing complexity toward the edges of the network whenever possible. Simulation results show that NBP effectively eliminates congestion collapse and that, when combined with ECSFQ, approximately max-min fair bandwidth allocations can be achieved for competing flows.

The emergence of multiprotocol label switching (MPLS) with its efficient support of explicit routing provides basic mechanisms for facilitating traffic engineering. Exploiting this capability of MPLS, we propose an adaptive multipath traffic engineering mechanism named LDM (load distribution over multipath). The main goal of LDM is to enhance the network utilization as well as the network performance by adaptively splitting traffic load among multiple paths. LDM takes pure dynamic approach not requiring any a priori traffic load statistics. Routing decisions are made at the flow level, and traffic proportioning reflects both the length and the load of a path. LDM also dynamically selects a few good label switched paths (LSPs) according to the state of the entire network. We use simulation to compare the performance of LDM with the performance of several representative dynamic load distribution approaches as well as the traditional static shortest path only routing. The numerical results show that LDM outperforms the compared approaches in both the blocking ratio as well as the performance of the accepted traffic flows.

In this paper, we propose and investigate a bandwidth-efficient multicast routing protocol for ad-hoc networks. The proposed protocol achieves low communication overhead, namely, it requires a small number of control packet transmissions for route setup and maintenance. The proposed protocol also achieves high multicast efficiency, namely, it delivers multicast packets to receivers with a small number of transmissions. In order to achieve low communication overhead and high multicast efficiency, the proposed protocol employs the following mechanisms: (1) on-demand invocation of the route setup and route recovery processes to avoid periodic transmissions of control packets, (2) a new route setup process that allows a newly joining node to find the nearest forwarding node to minimize the number of forwarding nodes, and (3) a route optimization process that detects and removes unnecessary forwarding nodes to eliminate redundant and inefficient routes. Our simulation results show that the proposed protocol achieves high multicast efficiency with low communication overhead compared with other existing multicast routing protocols, especially in the ease where the number of receivers in a multicast group is large.

The end-to-end nature of Internet congestion control is an important factor in its scalability and robustness. However, end-to-end congestion control algorithms alone are incapable of preventing the congestion collapse and unfair bandwidth allocations created by applications which are unresponsive to network congestion. In this paper, we propose and investigate a new congestion avoidance mechanism called Network Border Patrol (NBP). NBP relies on the exchange of feedback between routers at the borders of a network in order to detect and restrict unresponsive traffic flows before they enter the network. The NBP mechanism is compliant with the Internet philosophy of pushing complexity toward the edges of the network whenever possible. Simulation results show that NBP effectively eliminates congestion collapse, and that, when combined with fair queueing, NBP achieves approximately max-min fair bandwidth allocations for competing network flows.

Layered transmission of data is often recommended as a solution to the problem of varying bandwidth constraints in multicast video applications. Multilayered encoding, however, is not sufficient to provide high video quality and high network utilization, since bandwidth constraints frequently change over time. Adaptive techniques capable of adjusting the rates of video layers are required to maximize video quality and network utilization. We define a class of algorithms known as source-adaptive multilayered multicast (SAMM) algorithms. In SAMM algorithms, the source uses congestion feedback to adjust the number of generated layers and the bit rate of each layer. We contrast two specific SAMM algorithms: an end-to-end algorithm, in which only end systems monitor available bandwidth and report the amount of available bandwidth to the source, and a network-based algorithm, in which intermediate nodes also monitor and report available bandwidth. Using simulations that incorporate multilayered video codecs, we demonstrate that SAMM algorithms can exhibit better scalability and responsiveness to congestion than algorithms that are not source-adaptive. We also study the performance trade-offs between end-to-end and network-based SAMM algorithms.

In this paper, we propose and investigate a bandwidth-efficient multicast routing protocol for ad-hoc networks. The proposed protocol achieves low communication overhead, namely, it requires a small number of control packet transmissions for route setup and maintenance. The proposed protocol also achieves high multicast efficiency, namely, it can deliver multicast packets to receivers with a small number of multicast transmissions. To achieve high multicast efficiency with low communication overhead, the proposed protocol employs the following mechanisms: (1) on-demand invocation of the route setup and route maintenance process to avoid periodical control packet transmissions, (2) a new route setup mechanism that allows a newly joining node to find the nearest forwarding node, in order to minimize the number of added forwarding nodes, and (3) a route optimization mechanism that can detect and delete unnecessary forwarding nodes. Our simulation results show that the proposed protocol achieves high multicast efficiency with low communication overhead compared with other multicast routing protocols.

In ATM networks, the available bit rate (ABR) service enables data sources to efficiently utilize network resources without adversely affecting the performance of non-ABR, guaranteed traffic such as video. Most research to date on ABR performance has not used realistic assumptions. Unrealistic assumptions such as persistent data sources, an absence of non-ABR traffic, and inaccurate TCP models may lead to misleading results. We address the above shortcomings of previous research. In a comparison of two major ABR schemes, the rate based Explicit Rate Indication for Congestion Avoidance (ERICA) and the credit based Quantum Flow Control (QFC), we find that the performance may be significantly better with QFC than previously thought when bursty traffic sources and bursty interfering traffic is considered.

Recent measurements of packet networks have shown that packet traffic exhibits long-range-dependent property. Studies have also discovered that this property is not adequately captured by the conventional Markov traffic models and often yields queueing behavior such as heavy-tailed or subexponential queue length distribution. These findings have led to the implication that long-range-dependent traffic has a significant impact on determining queueing performance and dimensioning buffer requirement in general. This paper examines the impact of long-range-dependent traffic on providing connectionless services over ATM networks using AAL5 protocol. As connectionless data packets enter an ATM network, they are encapsulated into AAL5 frames. Within the ATM network, AAL5 frames are segmented into cells for transmission and reassembled at intermediate connectionless servers for frame switching. A queueing model characterizing the AAL5 frame reassembly operation is developed and analyzed to investigate the impact of long-range-dependent traffic on dimensioning the frame reassembly buffer. The buffer overflow probability is analytically derived. Tight upper bound and lower bound of the AAL5 frame loss probability are also established. The analysis shows that long-range-dependent traffic and conventional Markovian traffic yield similar queueing behavior in the frame reassembly operation.

Network architectures that can efficiently transport high quality, multicast video are rapidly becoming a basic requirement of emerging multimedia applications. The main problem complicating multicast video transport is variation in network bandwidth constraints. An attractive solution to this problem is to use an adaptive, multi-layered video encoding mechanism. We consider two such mechanisms for the support of video multicast; one is a rate-based mechanism that relies on explicit rate congestion feedback from the network, and the other is a credit-based mechanism that relies on hop-by-hop congestion feedback. The responsiveness, bandwidth utilization, scalability and fairness of the two mechanisms are evaluated through simulations. Results suggest that while the two mechanisms exhibit performance trade-offs, both are capable of providing a high quality video service in the presence of varying bandwidth constraints.

In the next generation high-speed wireless networks, it is important to provide quality-of-service (QoS) guarantees as they are expected to support multimedia applications. This paper proposes an admission control scheme based on adaptive bandwidth reservation to provide QoS guarantees for multimedia traffic carried in high-speed wireless cellular networks. The proposed scheme allocates bandwidth to a connection in the cell where the connection request originates and reserves bandwidth in all neighboring cells. When a user moves to a new cell and a handoff occurs, bandwidth is allocated in the new cell, bandwidth is reserved in the new cell's neighboring cells, and reserved bandwidth in more distant cells is released. The amount of bandwidth to reserve is dynamically adjusted, reflecting the current network conditions. The performance of the proposed scheme is evaluated through simulations of realistic cellular environments. The simulated network consists of a large number of cells, mobile users with various movement patterns are assumed, and a variety of multimedia applications (e.g., audio phone, video conference, video on demand, file transfer, etc.) is considered. It is shown that the proposed scheme provides small handoff dropping probability (i.e., the probability that handoff connections are dropped due to a lack of bandwidth) and achieves high bandwidth utilization.

The available bit rate (ABR) service dynamically utilizes free bandwidth in ATM networks for data traffic. ABR enables data sources to efficiently utilize network resources without adversely affecting the performance of guaranteed traffic such as for video. However, the majority of research to date on the performance of ABR schemes has ignored the effects of guaranteed traffic. The vast majority of research has also only utilized persistent sources although real data traffic, such as Internet traffic, is bursty. This paper presents a simulation study of bursty Internet TCP traffic over ATM with bursty interfering traffic streams. Two major ABR schemes, rate based ABR with explicit rate (ER) control and the credit based quantum flow control (QFC), are compared. The results indicate that performance with QFC is significantly better than with ER control.

While existing research shows that reactive congestion control mechanisms are capable of providing high video quality and channel utilization for point-to-point real-time video, there has been relatively little study of the reactive congestion control of point-to-multipoint video, especially in ATM networks. Problems complicating the provision of multipoint, feedback-based real-time video service include: (1) implosion of feedback returning to the source as the number of multicast destinations increases and (2) variance in the amount of available bandwidth on different branches in the multipoint connection. A new service architecture is proposed for real-time multicast video, and two multipoint feedback mechanisms to support this service are introduced and studied. The mechanisms support a minimum bandwidth guarantee and the best effort support of video traffic exceeding the minimum rate. They both rely on adaptive, multilayered coding at the video source and closed-loop feedback from the network in order to control both the high and low priority video generation rates of the video encoder. Simulation results show that the studied feedback mechanisms provide, at the minimum, a quality of video comparable to a constant bit rate (CBR) connection reserving the same amount of bandwidth. When unutilized network bandwidth becomes available, the mechanisms are capable of exploiting it to dynamically improve video quality beyond the minimum guaranteed level.

In this paper, object-oriented techniques and concepts (e.g., object migration inheritance delegation) and reflection are applied to the design of a distributed system called the reflective object-oriented distributed system (RODS). RODS hides object distribution and underlying platform heterogeneity to simplify the development of distributed multimedia applications. RODS has two major components: the flexible multimedia object adaptor (FMOA) and the reflective object request broker (RORB). The FMOA is an application framework of classes to facilitate the construction of distributed multimedia applications. The RORB is a CORBA object request broker that integrates additional features such as smart proxies, delegation, migration, and sharing. By utilizing a reflective model, RORB allows the application programmer and the system to dynamically modify distributed object management policies.

This paper analyzes an ATM switching node with correlated cell arrivals. An ATM switching node is modeled as a discrete-time finite-buffer queue. Cells arrive according to a semi-Markovian process, where the number of cell arrivals in a given slot depends on the state of the underlying (M-state) Markov chain in both the given slot and the previous slot. This paper presents analysis for various cell loss characteristics such as cell loss probability, the consecutive loss probability, and the distribution of loss period lengths. It also obtains characteristics of the cell output process from an ATM switching node including the joint distribution of successive cell interdeparture times, and the distributions of busy and idle periods of the cell output process. It is shown through numerical examples that both the correlation and the variation of cell arrivals significantly affect the cell loss and the output process characteristics.>

The backward compatibility dilemma introduced by the deployment of a high-speed, wide-area asynchronous transfer mode (ATM) network is examined. The problem is that emerging ATM network must be capable of transporting traffic sources such as digital voice and video, but many of the design choices that optimize ATM for these sources make the transport of traditional connectionless data traffic difficult. The overlay of a virtual datagram network of connectionless servers onto a public ATM network to provide connectionless service is proposed to solve the problem. This method provides low transfer delay and efficiently utilizes network resources. Architectural issues are described, and an implementation of a connectionless server is presented.>

The performance of an asynchronous transfer mode (ATM) switching node is analyzed, taking cell arrival correlation into consideration. An ATM switching node is modeled as a discrete-time finite-buffer queue. Cell arrivals are assumed to follow a semi Markovian process. Various characteristics of the cell loss and the distribution function of the cell output process from an ATM switching node are analyzed. The cell loss probability, the consecutive loss probability, the distribution of loss period lengths, the joint distribution of successive cell interdeparture times, and the distributions of busy and idle periods are determined. It is shown that both the correlation and the variation of cell arrivals significantly affect the cell loss and the output process characteristics.>

A framework containing a number of resources, languages, and tools for generating customized protocols that support diverse multimedia applications running on high-performance networks is described. The framework facilitates the configuration of application-tailored, function-based communication protocols that are automatically synthesized from high-level specifications.

Abstract Computer communication systems must undergo significant changes to keep pace with the increasingly demanding and diverse multimedia applications that will run on the next generation of high‐performance networks. To facilitate these changes, we are developing A Dynamically Assembled Protocol Transformation, Integration and evaluation Environment (ADAPTIVE). ADAPTIVE provides an integrated environment for developing and experimenting with flexible transport system architectures that support lightweight and adaptive communication protocols for diverse multimedia applications running on high‐performance networks. Our approach employs a collection of reusable ‘building‐block’ protocol mechanisms that may be composed together automatically based upon functional specifications. The resulting protocols execute in parallel on several target platforms including shared‐memory and message‐passing multiprocessors. ADAPTIVE provides a framework for (1) determining the functionality of customized lightweight protocol configurations that efficiently support multimedia applications and (2) mapping this functionality onto efficient parallel process architectures.

Transport system services that integrate operating system resources such as CPUs, virtual memory, and I/O devices with network protocols to support distributed multimedia applications running on local and wide area networks are described and classified. A taxonomy is presented, and four commercial and experimental transport systems are compared and evaluated in terms of their protocol processing support. The systems discussed are System V UNIX STREAMS, the BSD UNIX networking subsystem, the x-kernel, and the Choices Conduit system.>

This paper describes a high-performance transport system architecture called ADAPTIVE, 'A Dynamically Assembled Protocol Transformation, Integration, and Validation Environment'. The authors are developing ADAPTIVE to support multimedia applications running on high-speed networks. ADAPTIVE is a transformational system providing policies and mechanisms that automatically specify and configure a flexible adaptive transport system. In addition, it provides a controlled prototyping environment for monitoring, analyzing, and experimenting with the effects of different transport system designs and implementations on application performance.>

The authors consider a queuing system with a finite buffer and multiple heterogeneous arrival streams. They focus on Markov modulated arrival processes with different burstings and investigate the loss of individual arrival streams when the parameters of the heterogeneous arrival streams are varied. The analysis includes both continuous-time and discrete-time treatments of multiplexed heterogeneous Markov modulated arrivals. Loss probabilities are derived for a priority packet discarding scheme. A new characterization of an arrival stream is introduced, referred to as a self-loss, and it is used to qualitatively predict the effects of multiplexing bursty streams with nonbursty streams. The effectiveness of priority packet discarding is also investigated through numerical examples.>

The design and implementation of a local area network architecture based on collision avoidance, called the collision avoidance multiple broadcast (CAMB) tree network, are described. The implementation follows the protocol layering architecture of the IEEE 802 local area networks and includes CAMB tree switches, station/network interface boards, and support of transport protocols. The performance measurements of the experimental CAMB tree network are presented.>

The authors investigate the effects of protocol processing overhead on the performance of error recovery schemes. The focus is on the edge-to-edge error recovery scheme, in which retransmissions of erred packets only take place between source and destination nodes. An approximation is obtained for the Laplace transform for the distribution of the end-to-end packet transfer delay, considering the processing time required for error recovery. The performance of the link-by-link error recovery scheme, in which retransmissions take place between adjacent nodes, is evaluated and compared to the performance of the edge-to-edge scheme. Numerical results from a tandem queuing network model show that for a network with very-high-speed/low-error-rate channels, an edge-to-edge scheme gives a smaller packet transmission delay than a link-by-link scheme for both go-back-N and selective-repeat retransmission procedures, while keeping the packet loss probability sufficiently small.>

The authors survey a number of important research topics in ATM (asynchronous transfer mode) networks. The topics covered include mathematical modeling of various types of traffic sources, congestion-control and error-control schemes for ATM networks, and priority schemes to support multiple classes of traffic. Standard activity for ATM networks and future research problems in ATM are also presented. It is shown that the cell-arrival process for data sources can be modeled by a simple Poisson process. However, voice sources or video sources require more complex processes because of the correlation among cell arrivals. Due to the effects of high-speed channels, preventive control is more effective in ATM networks than reactive control. Due to the use of optical fibers in ATM networks, the channel error rate is very small. The effects of propagation delay and processing time become significant in such high-speed networks. These fundamental changes trigger the necessity to reexamine the error-control schemes used in existing networks. Due to the diversity of service and performance requirements, the notion of multiple traffic classes is required, and separate control mechanisms should be used according to the traffic classes. The priority scheme is shown to be an effective method to support multiple classes of traffic.>

The performance of a broadcast star network is analyzed, under the assumption of synchronous operation of a network. An exact analysis for a broadcast star network with an infinite station population, an exact analysis for a small network with a finite station population, and an approximate analysis for a large network with a finite station population are presented. In synchronous operation, the channel time is slotted, and stations transmit only at the beginning of a slot. The throughput and the distribution of transmission delays are obtained. It is shown through simulations that a broadcast star operating under synchronous mode yields better performance than when operating under asynchronous mode, where transmissions of packets are not confined to the beginning of slots and stations start transmission at any time.>

The major ATM switch architectures and their improvement techniques are categorized and discussed, focusing on performance issues. The performance measures of interest are the maximum throughput, the delay time, and the cell loss probability. The major assumptions and notations used are summarized. A survey is presented of nonblocking switches and their improvement techniques. The performances of a variety of nonblocking switches are compared.>

Presents a design of a collision avoidance switch for the collision avoidance multiple broadcast (CAMB) tree using transistor-transistor logic (TTL) devices. A simulation study exploring the performance of the broadcast star network in both synchronous and asynchronous operations is also presented. Simulation results show that the slotted broadcast star achieves better performance than the unslotted broadcast star, especially when the traffic load is heavy.>

The performance of an asynchronous transfer mode (ATM) network subject to admission control and traffic smoothing is analyzed. Basically, an ATM switch is modeled as a discrete-time single-server queuing system in which a new call joins existing calls. Cell arrivals from a new call are assumed to follow a general distribution. It is also assumed that aggregated arrivals of cells from existing calls form batch arrivals with a general batch size distribution and a geometric distribution of the interarrival times between batches. Both finite- and infinite-buffer cases are considered. An exact analysis yields the waiting time distribution and cell loss probability for a new call and for existing calls. Numerical examples are given to show how the network performance depends on the statistics of a new call (burstiness, time that a call stays in an active or inactive state, etc.) and to demonstrate the effectiveness of admission control and traffic smoothing.>

A message-driven model for function-free Horn logic is presented, where the knowledge base is represented as a network of logical processing elements communicating with one another exclusively through messages. The lack of centralized control and centralized memory makes this model suitable for implementation on highly parallel asynchronous computer architecture. The performance of this message-driven system is analyzed and compared to that of a sequential resolution scheme using backtracking. Closed-form expressions for both approaches are derived.>

An analysis is made of the throughput and delay performance of two classes of free-access tree algorithms with minislots. In one class, binary feedback information is available in minislots, and in the other, ternary feedback information is available. It is shown that the highest maximum throughput 0.56714 is achieved in the limiting case where the number of minislots in a (large) slot is infinity and minislot overhead is zero. A lower bound of the average transmission delays in these algorithms is analytically derived. The obtained lower bound is also a lower bound of the average delay of the whole class of the free-access algorithms.>

The design is presented of a broadcast star network that uses a collision-avoidance switch. This network combines the benefits of random access (low delay when traffic is light; simple, distributed, and therefore robust protocols) with excellent network utilization. Performance analyses of a slotted broadcast star network are then presented to obtain distributions of the transmission delay: an exact analysis for a network with an infinite number of stations, and an approximate analysis for a network with a finite number of stations.>

The performance of a token-passing ring network with packetized voice/data mixed traffic is investigated through extensive simulations. Both data and voice users are modeled in the simulations. Data users produce bursty traffic. Voice traffic is modeled as having alternating talkspurts and silences, with generation of voice packets at a constant rate during talkspurts and no packet generation during silence periods. Token passing ring local area networks are shown to effectively handle both voice and data traffic. The effects of system parameters (e.g. voice packet length, talkspurt/silence lengths, data traffic intensity, and limited exhaustive service disciplines) on network performance are discussed.>

A discussion is presented of the robustness of the Capetanakis collision resolution algorithm for multiple-access networks. In particular, they show that the algorithm is robust against errors that cause different stations to see different feedback information on the results of the transmission attempts, and they quantify, using simulation, the performance (throughput/delay) degradation caused by those unshared errors.>

An investigation is made of various error recovery schemes for a very-high-speed packet switched network. The schemes investigated are: (1) edge-to-edge recovery, whereby retransmission of erred packets only takes place between source and destination nodes, (2) link-by-link recovery, whereby retransmission only takes place between adjacent switching nodes, and (3) link-edge-combined recovery, using both link-by-link and edge-to-edge schemes. The performance measures are the distribution of transmission delay and error probability of packets across a network. To obtain these measures, the authors develop a tandem queuing network model with feedbacks, where each queue represents a protocol layer within a switching node rather than a switching node as a whole. Numerical results show that for a network with very-high-speed/low-error-rate channels, the edge-to-edge scheme gives the smallest packet transmission delay, while keeping the packet error probability sufficiently small. The only drawback of the edge-to-edge scheme is that it causes network saturation at smaller network traffic load than the older schemes.>

Packet collisions and their resolution create a performance bottleneck in random access LANs. A hardware solution to this problem is to use collision avoidance switches. These switches allow the implementation of random access protocols without the penalty of collisions among packets. We describe the designs of some tree LANs that use collision avoidance switches. The collision avoidance LANs we describe are broadcast star and CAMB tree (Collision Avoidance Multiple Broadcast tree). We next present a possible implementation of a collision avoidance switch using currently available photonic devices. Finally, we show the performance of broadcast star and CAMB tree networks using simulations.

Introduction of the packet switching technique into digitized voice communication may afford great advantages in efficient use of the channel, compared to both circuit-switched and DSI systems. Detailed characteristics, however, have not been obtained because of difficulty in the exact analysis. Hence, simalation models are developed in this paper for the packetized voice transmission system, and various characteristics such as tranmission delays and loss probability of voice packets are obtained. We further evaluate three types of voice packet reassembly strategy at the receiving terminal, and obtain the optimal packet length, which keeps both overall packet transmission delay and packet loss probabilty less than a certain permissible value. Comparison among three strategies is also stated.

A method for realizing a circuit and packet integrated access scheme in a satellite communication channel is considered. Two kinds of terminals are assumed, namely, bursty terminals for handling bursty traffic and heavily loaded terminals for long-holdingtime message traffic. In this method, the channel frame is divided into two subframes: one is for bursty terminals, and the other is for heavily loaded terminals. The subframe for heavily loaded terminals is further divided into two subchannels, a reservation subchannel (consisting of small slots) and a message subchannel. The bursty terminals transmit their packets in their dedicated subframes on the slotted ALOHA protocol. The heavily loaded terminal having a message transmits, first of all, a reservation packet in a randomly selected small slot of the reservation subchannel to reserve slots in the coming message subchannels. One slot in the same position of each of the succeeding message subchannels is reserved for the terminal until the end-of-use flag, transmitted from the terminal, is received by the satellite. Mean transmission delays for both kinds of traffic in this method are analytically obtained. We show that there exists an optimal frame length which minimizes mean transmission delay for one kind of traffic while keeping mean transmission delay for the other kind under some permissible value.

This paper considers a satellite communication system with the slotted ALOHA access mode where the satellite is assumed to have on-board regeneration ability and the finite buffer capacity for storing noncollided packets from the uplink. The system is analyzed with the downlink bandwidth restricted to1/Lof the uplink bandwidth, and with the conventional slotted ALOHA system whereL = 1and the buffer size at the satellite is equal to zero. Maximum throughput and throughput-delay characteristics of this system are analytically obtained. The satellite is given an optimal buffer size to achieve maximum throughput and minimum packet transmission delay. It is shown that the constraints that lead to the downlink bandwidth being lower than the uplink bandwidth do not necessarily result in throughput penalties in a buffered satellite system. Furthermore, when the total bandwidth which can be divided between up- and downlinks is given, if two or three times that of the downlink is assigned to the uplink and if the satellite has buffer storage space only for a few packets, we show that maximum throughput of the system will be improved about 40 percent with little change in delay characteristics in comparison with that of the conventional slotted ALOHA system where the given total bandwidth is equally divided between up- and downlinks.