Congestion-free communication in high-speed packet networks

The process of packet clustering in a network with well-regulated input traffic is studied and a strategy for congestion-free communication in packet networks is proposed. The strategy provides guaranteed services per connection with no packet loss and an end-to-end delay which is a constant plus a small bounded jitter term. It is composed of an admission policy imposed per connection at the source node, and a particular queuing scheme practiced at the switching nodes, which is called stop-and-go queuing. The admission policy requires the packet stream of each connection to possess a certain smoothness property upon arrival at the network. This is equivalent to a peak bandwidth allocation per connection. The queuing scheme eliminates the process of packet clustering and thereby preserves the smoothness property as packets travel inside the network. Implementation is simple     

Modeling and measurement techniques in packet communication networks

Considerable advances in the modeling and measurements of packet-switched networks have been made since this concept emerged in the late sixties. In this paper, we first review the modeling techniques that are most frequently used to study these packet transport networks; for each technique we provide a brief introduction, a discussion of its capabilities and limitations, and one or more representative applications. Next we review the basic measurement tools, their capabilities, their limitation, and their applicability to and implementation in different networks, namely land based wire networks, satellite networks, and ground packet radio networks; we also show the importance of well-designed experiments in satisfying the many measurement goals. Finally we discuss briefly some open problems for future research.     

Topological analysis of packet networks

The author describes a unified approach for the topological analysis of nonhierarchical and hierarchical packet networks. The approach differs from previous approaches in adopting an end-to-end mean delay objective and including a variety of practical routing constraints. These include limits on the number of paths allowed in a route, limits on the number of hops allowed in a path, and constraints due to prevalent virtual circuit implementations. For a broad range of networks, quantitative analysis based on this approach provides new insights into the complex relationships between network topology and routing and delay constraints. It is shown that the sole use of a network average delay criterion often leads to network designs that exhibit poor end-to-end mean delays for some node pairs, and that it is possible to configure networks that meet an end-to-end mean delay objective for every node pair at little or no additional cost     

Performance analysis of data packet discarding in ATM networks

Data performance in ATM networks should be measured on the packet level instead of the cell level, since one or more cell losses within each packet is equivalent to the loss of the packet itself. Two packet-level control schemes, packet tail discarding and early packet discarding, were proposed to improve data performance. In this paper, a new stochastic modeling technique is developed for performance evaluation of two existing packet-discarding schemes at a single bottleneck node. We assume that the data arrival process is independent of the nodal congestion, which may represent the unspecified bit-rate traffic class in ATM networks, where no end-to-end feedback control mechanism is implemented. Through numerical study, we explore the effects of buffer capacity, control threshold, packet size, source access rate, underlying high-priority real-time traffic, and loading factor on data performance, and discuss their design tradeoffs. Our study shows that a network system can he entirely shut down in an overload period if no packet-discarding control scheme is implemented, under the assumption that there are no higher layer congestion avoidance schemes. Further, unless with sufficiently large buffer capacity, early packet discarding (EPD) always outperforms packet tail discarding (PTD) significantly under most renditions. Especially under the overload condition, EPD can always achieve about 100% goodput and 0% badput, whereas the PTD performance deteriorates rapidly. Among all the factors, the packet size has a dominant impact on EPD performance. The optimal selection of the EPD queue control threshold to achieve the maximum goodput is found to be relatively insensitive to traffic statistics     

Modeling and performance analysis of multihop packet radio networks

The design of packet radio networks involves a large number of issues which interact in a very complex fashion. Many of these pertain to the RF channel and its use, others pertain to the operational protocols. Clearly, no single model can be formulated which incorporates all the necessary parameters and leads to the optimum solution. The one essential element which complicates matters is that, contrary to point-to-point networks in which each channel is utilized by a single pair of nodes, the radio channel in packet radio networks is a multiaccess broadcast resource: i) in a given locality determined by radio connectivity, the channel is shared by many contending users, hence the need for channel access protocols; ii) radio is a broadcast medium and thus the action taken by a node has an effect on the actions taken by neighboring nodes and their outcome. Despite the complexity of the problem, there has been significant progress worth reporting on. The work accomplished so far has been either the analysis of specific examples of networks or an attempt to create models that would be useful in the design of general networks. The purpose of this paper is to survey the various modeling techniques that have been used for the performance analysis of packet radio networks, and to discuss the assumptions underlying these models, their scope of applicability, and some of the results obtained.     

Multiple-access protocols for satellite packet communication networks: A performance comparison

This paper considers satellite packet communication networks with a large population of bursty users and presents an analytic comparison of the throughput versus average message delay trade-off characteristics of multiple-access protocols. The following six multiple-access protocols are examined: 1) slotted ALOHA, 2) reservation-ALOHA, 3) a reservation protocol with a slotted ALOHA reservation channel, 4) a reservation protocol with a TDMA reservation channel, 5) SRUC (Split Reservation Upon Collision), and 6) fixed assigned TDMA. All the protocols are required to ensure that all packets of a message are correctly received in the proper order at the destination. Then, a unified presentation of the delay-throughput performance of the protocols is given by means of an analytical technique called equilibrium point analysis. The throughput versus average message delay tradeoff characteristics are compared taking into account the system stability.     

Adaptive rate controlled,robust video communication over packet wireless networks

Video transmission over wireless packet networks is gaining importance due to the concept of universal personal communication. Further, it is considered an important step towards wireless multimedia. The challenge however is to achieve good video quality over mobile channels, where typically the channel conditions vary due to signal fading. Hence this paper investigates adaptive rate controlled video transmission for robust video communication under packet wireless environment. A combination of mobile and an ATM backbone network is assumed in this work. An error resilient design for the video coder, as proposed in Rajugopal et al. (1996) is employed here. This video coder comprises wavelet transform (WT), multi-resolution motion estimation (MRME) and a robust design for zero tree quantization. Two configurations, one employing MRME and the other using 1D-WT for temporal analysis, are considered for the video coder. Adaptive dynamic rate control is required to adapt the video communication to the channel conditions. It provides more channel protection when the channel is severe and improves the source rate and hence the performance when the conditions are favorable. An algorithm for dynamic rate control under varying channel conditions is proposed in this paper. It is evaluated under narrowband and broadband channel conditions. From the results, it is concluded that the dynamic rate control is very effective in optimizing the quality under varying mobile channel conditions. It was observed that the dynamic rate control provides at least an acceptable video quality under severe channel conditions and a good video quality when the channel conditions are favorable. This revised version was published online in July 2006 with corrections to the Cover Date.     

Energy-based interference analysis of heterogeneous packet radio networks

While the use of radio technology for wireless data communications has increased rapidly, the wide variety of radio interfaces being used has made interference investigations hard to perform. With that in mind, we present a novel approach for analyzing packet radio communications, applicable to interfering heterogeneous networks, which leads to tractable analytical expressions. The core of the approach is an analytical framework modeling each network with individual properties for the packet types and the channel sets used, while taking path loss between all network nodes into account. Furthermore, we present a derivation of closed-form expressions for the throughput of the networks, thus allowing for the investigation of important mechanisms limiting network and system performance. The expressions enable fast and flexible analysis to be performed without extensive computer simulations or measurement campaigns. To illustrate the use of the framework and the strength of the closed-form expressions, we analyze a heterogeneous example system consisting of one IEEE 802.11b network and multiple Bluetooth networks that use multiple packet types. In the analysis, we also take the adjacent channel interference into account when calculating network throughput as functions of the number of interferers in the system.     

Packet-error probability analysis for FH-CDMA unslotted packet networks

This letter is concerned with the interference analysis in time-unslotted frequency-hopping code-division multiple-access packet networks with error control coding. We derive, under the conditional independence assumption, a new exact closed-form expression for the packet error probability of a reference packet in the presence of other K interfering packets. In contrast to the previously known results, the computational complexity of the new result is independent of K. We also develop a new tight upper bound on packet error probability relying on mapping the unslotted system into an equivalent slotted one experiencing the same level of short-term average interference. The accuracy of the proposed approximation is demonstrated by some numerical examples and is shown to be tighter than the previously used results.     

Self-healing in wideband packet networks

A brief overview is given of wideband packet technology (WPT), with special attention to the features affecting the feasibility of self-healing, a method for automatic failure recovery proposed herein. The objectives of self-healing are set forth, and the failure detection strategy is described. With this strategy, nondisruptive trunk monitoring (self-monitoring) at the data link layer of an open system interconnection (OSI) reference model is performed to detect facility failure of under 1 s. The bit error rate (BER) on a transmission link is estimated on the basis of counts of the number of packets with cyclic redundancy check (CRC) errors at the data link layer. This failure detection strategy can respond to link failures for any given traffic mix by the provisioning of duration and level of unacceptable sustained BER. The mechanisms used to achieve recovery in frame-relay networks using packet streams to carry users' traffic are discussed. Provisioning of the WPT nodes and the network is addressed     

Providing deterministic packet delays and packet losses in multimedia wireless networks

‘Anytime, anywhere’ communication, information access and processing are much cherished in modern societies because of their ability to bring flexibility, freedom and increased efficiency to individuals and organizations. Wireless communications, by providing ubiquitous and tetherless network connectivity to mobile users, are therefore bound to play a major role in the advancement of our society. Although initial proposals and implementations of wireless communications are generally focused on near‐term voice and electronic messaging applications, it is recognized that future wireless communications will have to evolve towards supporting a wider range of applications, including voice, video, data, images and connections to wired networks. This implies that future wireless networks must provide quality‐of‐service (QoS) guarantees to various multimedia applications in a wireless environment. Typical traffic in multimedia applications can be classified as either Constant‐Bit‐Rate (CBR) traffic or Variable‐Bit‐Rate (VBR) traffic. In particular, scheduling the transmission of VBR multimedia traffic streams in a wireless environment is very challenging and is still an open problem. In general, there are two ways to guarantee the QoS of VBR multimedia streams, either deterministically or statistically. In particular, most connection admission control (CAC) algorithms and medium access control (MAC) protocols that have been proposed for multimedia wireless networks only provide statistical, or soft, QoS guarantees. In this paper, we consider deterministic QoS guarantees in multimedia wireless networks. We propose a method for constructing a packet‐dropping mechanism that is based on a mathematical framework that determines how many packets can be dropped while the required QoS can still be preserved. This is achieved by employing: (1) An accurate traffic characterization of the VBR multimedia traffic streams; (2) A traffic regulator that can provide bounded packet loss and (3) A traffic scheduler that can provide bounded packet delay. The combination of traffic characterization, regulation and scheduling can provide bounded loss and delay deterministically. This is a distinction from traditional deterministic QoS schemes in which a 0% packet loss are always assumed with deterministically bounding the delay. We performed a set of performance evaluation experiments. The results will demonstrate that our proposed QoS guarantee schemes can significantly support more connections than a system, which does not allow any loss, at the same required QoS. Moreover, from our evaluation experiments, we found that the proposed algorithms are able to out‐perform scheduling algorithms adopted in state‐of‐the‐art wireless MAC protocols, for example Mobile Access Scheme Based on Contention and Reservation for ATM (MASCARA) when the worst‐case traffic is being considered. Copyright © 2002 John Wiley & Sons, Ltd.     

Queuing analysis of the selective repeat automatic repeat requestprotocol wireless packet networks

This paper presents an analytical approach for analyzing the mean packet delay and mean queue length at the transmitting terminal in wireless packet networks using the selective repeat (SR) automatic repeat request (ARQ) scheme to control the errors introduced by the nonstationary transmission channel. Each transmitting terminal is modeled as a discrete time queue with an infinite buffer. The nonstationary transmission channel is modeled as a two-state Markov chain. Comparisons of numerical predictions and simulation results are presented to highlight the accuracy of the proposed analytical approach     

Fluid analysis of delay and packet discard performance for QoSsupport in wireless networks

Providing quality-of-service (QoS) guarantees over wireless links requires thorough understanding and quantification of the interactions among the traffic source, the wireless channel, and the underlying link-layer error control mechanisms. We account for such interactions in an analytical model that we use to investigate the delay distribution and the packet discard rate (PDR) over a wireless link. Our analysis accommodates the inherent autocorrelations in both the traffic source as well as the channel error characteristics. An on-off fluid process is used to model the arrival of packets at the transmitter. These packets are temporarily stored in a first-in-first-out (FIFO) buffer before being transmitted over a channel with a time-varying and autocorrelated service rate. Using fluid analysis, we first derive the distribution for the queueing delay at the transmitter. As part of this analysis, we solve a fundamental fluid problem, namely, the probability distribution for the workload generated by a two-state fluid source over a fixed time interval. We then use the delay analysis to derive the PDR at the receiver. A closed-form expression for the effective bandwidth subject to a delay constraint is provided as a function of the source, channel, and error scheme parameters. This expression enables fast assessment of the bandwidth requirement of real-time traffic over QoS-based wireless networks. Numerical results and simulations are used to verify the adequacy of the analysis and to study the interactions among various system parameters     

Markov-based channel characterization for tractable performance analysis in wireless packet networks

Finite-state Markov chain (FSMC) models have often been used to characterize the wireless channel. The fitting is typically performed by partitioning the range of the received signal-to-noise ratio (SNR) into a set of intervals (states). Different partitioning criteria have been proposed in the literature, but none of them was targeted to facilitating the analysis of the packet delay and loss performance over the wireless link. In this paper, we propose a new partitioning approach that results in an FSMC model with tractable queueing performance. Our approach utilizes Jake's level-crossing analysis, the distribution of the received SNR, and the elegant analytical structure of Mitra's producer-consumer fluid queueing model. An algorithm is provided for computing the various parameters of the model, which are then used in deriving closed-form expressions for the effective bandwidth (EB) subject to packet loss and delay constraints. Resource allocation based on the EB is key to improving the perceived capacity of the wireless medium. Numerical investigations are carried out to study the interactions among various key parameters, verify the adequacy of the analysis, and study the impact of error control parameters on the allocated bandwidth for guaranteed packet loss and delay performance.     

A deterministic approach to the end-to-end analysis of packet flowsin connection-oriented networks

We analyze the worst-case behavior of general connection-oriented networks, with first-in-first-out (FIFO) queueing policy, forwarding packets along an arbitrary system of routes. A worst-case bound is proven for the end-to-end queueing delay and buffer size needed to guarantee loss-free packet delivery, given that sources satisfy a given source rate condition. The results are based on a novel deterministic approach and help in reconciling the discrepancy between the unstable worst-case behavior of FIFO-based networks and their good practical performance     

Queueing analysis of scheduling policies in copy networks ofspace-based multicast packet switches

Space-based multicast switches use copy networks to generate the copies requested by the input packets. In this paper our interest is in the multicast switch proposed by Lee (1988). The order in which the copy requests of the input ports are served is determined by the copy scheduling policy and this plays a major part in defining the performance characteristics of a multicast switch. In any slot, the sum of the number of copies requested by the active inputs of the copy network may exceed the number of output ports and some of the copy requests may need to be dropped or buffered. We first propose an exact model to calculate the overflow probabilities in an unbuffered Lee's copy network. Our exact results improve upon the Chernoff bounds on the overflow probability given by Lee by a factor of more than 10. Next, we consider buffered inputs and propose queueing models for the copy network for three scheduling policies: cyclic service of the input ports with and without fanout splitting of copy requests and acyclic service without fanout splitting. These queueing models obtain the average delay experienced by the copy requests. We also obtain the sustainable throughput of a copy network, the maximum load that can be applied to all the input ports without causing an unstable queue at any of the inputs, for the scheduling policies mentioned above     

Teletraffic analysis of hierarchical cellular communication networks

The sustained increase of users and the request for advanced multimedia services are amongst the key motivations for designing new high-capacity cellular telecommunication systems. The proposals that are being pursued by several studies and field implementations consider hierarchical architectures and dynamic resource allocation. A hierarchical cellular communication network is analyzed, taking user mobility into account and exploiting dynamic channel-allocation schemes. In particular, a finite number of users has been considered, moving at different speeds in a geographical region covered by a finite number of cells structured in two hierarchical levels: micro- and macrocells. For such a system, mobility and traffic models have been developed, both based on queueing networks analyzing maximum packing (MP), a dynamic channel-allocation scheme. The obtained results, validated by simulation experiments, allow the evaluation of main system-performance parameters in terms of new-call and handoff blocking probabilities, and forced-termination probability as a function of load and system parameters.