We propose an analytical model for IEEE 802.11 wireless local area networks (WLANs). The analytical model uses macroscopic descriptions of the distributed coordination function (DCF): the backoff process is described by a few macroscopic states (medium-idle, transmission, and medium-busy), which obviates the need to track the specific backoff counter/backoff stages. We further assume that the transitions between the macroscopic states can be characterized as a continuous-time Markov chain under the assumption that state persistent times are exponentially distributed. This macroscopic description of DCF allows us to utilize a two-dimensional continuous-time Markov chain for simplifying DCF performance analysis and queueing processes. By comparison with simulation results, we show that the proposed model accurately estimates the throughput performance and average queue length under light, heavy, or asymmetric traffic.

Video traffic occupies a major part of current mobile traffic. The characteristics of video traffic are dominated by the behavior of the video application users. This paper uses a state transition diagram to analyze the behavior of video application users on smart phones. Video application users are divided into two categories; keyword search users and initial screen users. They take different first action in video viewing. The result of our analysis shows that the patience of video application users depends on whether they have a specific purpose when they launch a video application or not. Mobile network operators can improve the QoE of video application users by utilizing the results of this study.

In NGN standards, End Host, also referred to as Terminal Equipment (TE), holds an important place in end-to-end path performance. However, most researchers neglect TE performance when considering performance of end-to-end paths. As far as the authors' knowledge goes, no previous study has proposed a model for TE performance. This paper proposes a method for measuring performance of TE and model extraction based on measurement data. The measurement was made possible with the use of a special NPU (Network Processing Unit) implemented as a programmable NIC. Along with the probing itself, a framework for removing the skew between the NPU and OS is developed in this paper. The multidimensional analysis includes method of probing, packet size and background traffic volume, and studies their effect on TE performance. A method for extracting a generic TE model is proposed. The outcome of this research can be used for modelling TE in simulations and in modelling end-to-end performance when considering QoS in NGN.

Multipath routing has been extended to Border Gateway Protocol (BGP), the current de facto inter-domain routing protocol, to address the reliability and performance issues of the current Internet. However, inter-domain multipath routing introduces a significant challenge for scalability due to the large scale of the inter-domain routing system. At the same time it also introduces new challenges in terms of security and security related overhead. In this paper, we propose a regional multipath approach, Regional Multipath Inter-domain Routing (RMI), where multiple paths are only allowed to be propagated within a well-defined range. With multipath routing in a region, we enable inter-domain routing with rich path diversity and improved security, and no longer have to sacrifice scalability. We show how to propagate multiple paths based on the region by theoretical analysis and by extensive simulations. Our simulations show that the number of messages generated using this approach and the convergence delay are much less than those of BGP and BGP with full multipath advertisement.

Interconnection networks play a crucial role in the performance of massively parallel computers. Hierarchical interconnection networks provide high performance at low cost by exploring the locality that exists in the communication patterns of massively parallel computers. A Tori connected Torus Network (TTN) is a 2D-torus network of multiple basic modules, in which the basic modules are 2D-torus networks that are hierarchically interconnected for higher-level networks. This paper addresses the architectural details of the TTN and explores aspects such as node degree, network diameter, cost, average distance, arc connectivity, bisection width, and wiring complexity. We also present a deadlock-free routing algorithm for the TTN using four virtual channels and evaluate the network's dynamic communication performance using the proposed routing algorithm under uniform and various non-uniform traffic patterns. We evaluate the dynamic communication performance of TTN, TESH, MH3DT, mesh, and torus networks by computer simulation. It is shown that the TTN possesses several attractive features, including constant node degree, small diameter, low cost, small average distance, moderate (neither too low, nor too high) bisection width, and high throughput and very low zero load latency, which provide better dynamic communication performance than that of other conventional and hierarchical networks.

Three-dimensional (3D) wafer stacked implementation (WSI) has been proposed as a promising technology for massively parallel computers. A hierarchical 3D-torus (H3DT) network, which is a 3D-torus network of multiple basic modules in which the basic modules are 3D-mesh networks, has been proposed for efficient 3D-WSI. However, the restricted use of physical links between basic modules in the higher level networks reduces the dynamic communication performance of this network. A torus network has better dynamic communication performance than a mesh network. Therefore, we have modified the H3DT network by replacing the 3D-mesh modules by 3D-tori, calling it a Modified H3DT (MH3DT) network. This paper addresses the architectural details of the MH3DT network and explores aspects such as degree, diameter, cost, average distance, arc connectivity, bisection width, and wiring complexity. We also present a deadlock-free routing algorithm for the MH3DT network using two virtual channels and evaluate the network's dynamic communication performance under the uniform traffic pattern, using the proposed routing algorithm. It is shown that the MH3DT network possesses several attractive features including small diameter, small cost, small average distance, better bisection width, and better dynamic communication performance.

A principal attraction of ATM networks, in both wired and wireless realizations, is that the key quality of service (QoS) parameters of every call, including end-to-end delay, jitter, and loss are guaranteed by the network when appropriate cell-level traffic controls are imposed at the user network interface (UNI) on a per call basis, utilizing the peak cell rate (PCR) and the sustainable cell rate (SCR) values for the multimedia--voice, video, and data, traffic sources. There are three practical difficulties with these guarantees. First, while PCR and SCR values are, in general, difficult to obtain for traffic sources, the typical user-provided parameter is a combination of the PCR, SCR, and the maximum burstiness over the entire duration of the traffic. Second, the difficulty in accurately defining PCR arises from the requirement that the smallest time interval must be specified over which the PCR is computed which, in the limit, will approach zero or the network's resolution of time. Third, the literature does not contain any reference to a scientific principle underlying these guarantees. Under these circumstances, the issue of providing QoS guarantees in the real world, through traffic controls applied on a per call basis, is rendered uncertain. This paper adopts a radically different, high level approach to the issue of QoS guarantees. It aims at uncovering through systematic experimentation a relationship, if any exists, between the key high level user traffic characteristics and the resulting QoS measures in a realistic operational environment. It may be observed that while each user is solely interested in the QoS of his/her own traffic, the network provider cares for two factors: (1) Maximize the link utilization in the network since links constitute a significant investment, and (2) ensure the QoS guarantees for every user traffic, thereby maintaining customer satisfaction. Based on the observations, this paper proposes a two-phase strategy. Under the first phase, the average "link utilization" computed over all the links in a network is maintained within a range, specified by the underlying network provider, through high level call admission control, i.e. by limiting the volume of the incident traffic on the network, at any time. The second phase is based on the hypothesis that the number of traffic sources, their nature--audio, video, or data, and the bandwidth distribution of the source traffic, admitted subject to a specific chosen value of "link utilization" in the network, will exert a unique influence on the cumulative delay distribution at the buffers of the representative nodes and, hence, on the QoS guarantees of each call. The underlying thinking is as follows. The cumulative buffer delay distribution, at any given node and at any time instant, will clearly reflect the cumulative effect of the traffic distributions of the multiple connections that are currently active on the input links. Any bounds imposed on the cumulative buffer delay distribution at the nodes of the network will also dominate the QoS bounds of each of the constituent user traffic. Thus, for each individual traffic source, the buffer delay distributions at the nodes of the network, obtained for different traffic distributions, may serve as its QoS measure. If the hypothesis is proven true, in essence, the number of traffic sources and their bandwidth distribution will serve asa practically realizable high level traffic control in providing realistic QoS guarantees for every call. To verify the correctness of the hypothesis, an experiment is designed that consists of a representative ATM network, traffic sources that are characterized through representative and realistic user-provided parameters, and a given set of input traffic volumes appropriate for a network provider approved link utilization measure. The key source traffic parameters include the number of sources that are incident on the network and the constituent links at any given time, the bandwidth requirement of the sources, and their nature. For each call, the constituent cells are generated stochastically, utilizing the typical user-provided parameter as an estimate of the bandwidth requirement. Extensive simulations reveal that, for a given link utilization level held uniform throughout the network, while the QoS metrics--end-to-end cell delay, jitter, and loss, are superior in the presence of many calls each with low bandwidth requirement, they are significantly worse when the network carries fewer calls of very high bandwidths. The findings demonstrate the feasibility of guaranteeing QoS for each and every call through high level traffic controls. As for practicality, call durations are relatively long, ranging from ms to even minutes, thereby enabling network management to exercise realistic controls over them, even in a geographically widely dispersed ATM network. In contrast, current traffic controls that act on ATM cells at the UNI face formidable challenge from high bandwidth traffic where cell lifetimes may be extremely short, in the range of µs. The findings also underscore two additional important contributions of this paper. First, the network provider may collect data on the high level user traffic characteristics, compute the corresponding average link utilization in the network, and measure the cumulative buffer delay distributions at the nodes, in an operational network. The provider may then determine, based on all relevant criteria, a range of input and system parameters over which the network may be permitted to operate, the intersection of all of which may yield a realistic network operating point (NOP). During subsequent operation of the network, the network provider may guide and maintain the network at a desired NOP by exercising control over the input and system parameters including link utilization, call admittance based on the requested bandwidth, etc. Second, the finding constitutes a vulnerability of ATM networks which a perpetrator may exploit to launch a performance attack.

This paper considers the problem of providing relative service differentiation in IEEE 802.11 Wireless LAN by using different Medium Access Control (MAC) parameters for different service classes. We present an analytical model which predicts the saturation throughput of IEEE 802.11 Distributed Coordination Function with multiple classes of service. This model allows us to show that relative service differentiation can be achieved by varying the initial contention window alone. In this case, the saturation throughput of a station can be shown to be approximately inversely proportional to the initial contention window size being used by that station. The simulation results validate our analytical model.

Interconnection networks usually suffer from Little's Law: low cost implies low performance and high performance is obtained high cost. However, hierarchical interconnection networks provide high performance at low cost by exploring the locality that exists in communication patterns of massively parallel computers. In this paper, we propose a new hierarchical interconnection network, called Hierarchical Torus Network (HTN). This network reduces the number of vertical links in 3D stacked implementation while maintaining good network features. This paper addresses the architectural details of the HTN, and explores aspects such as the network diameter, average distance, bisection width, peak number of vertical links, and VLSI layout area of the HTN as well as for several commonly used networks for parallel computers. It is shown that the HTN possesses several attractive features including small diameter, small average distance, small number of wires, a particularly small number of vertical links, and economic layout area.

We investigate the capacity limitations of a WDM ring fiber-radio backbone incorporating wavelength interleaving where each base station drives a sectorized antenna interface. We also investigate the issues related to the merging of such networks with standard WDM infrastructures. The investigations show that re-allocating the interleaved WDM channels to fit within a 100 GHz block enables the millimeter-wave (mm-wave) fiber-radio system with sectorized antenna interfaces to integrate easily with WDM systems. The performance of a variety of channel allocations for the merged fiber-radio network is examined and simulation studies of the transmission of multiple channels are carried out. The overall network capacity of the merged mm-wave fiber-radio network is improved with the proposed channel allocation schemes.

This paper deals with network performance of 64kbit/s-based services supported in an ATM network and an ATM interworking network with 64kbit/s-based networks. It first clarifies network performance issues giving a model and objectives of study and experiment. A result of computational analysis is then presented, where a cell loss ratio of an order of 10-4 or 10-5 is obtained to give a performance equivalent to that currently used as objective values in existing networks for a 64kbit/s digital level. In order to capture the impact of cell loss and associated performance for application levels, an experimental test has been carried out using typical applications. Test results show that the cell loss ratio needs to be approximately ten times better than the bit error ratio for comparable performance for application levels. A cell loss ratio of better than 10-5, or an order of 10-6 considering an interworking situation, seems to be necessary according to the test results. It is further clarified by the test that a single cell is more valuable than a multiplexed cell for providing better cell loss resilience characteristics. Although not all applications of 64kbit/s networks have been tested, it is expected that the test results can be used as guidance for considering the support of 64 kbit/s services in an ATM and interworking networks.