Overlay networks which are dynamically created over underlying IP networks are becoming widely used for delivering multimedia contents since they can provide several additional user-definable services. Multiple overlay paths between a source-destination overlay node pair are designed to improve service robustness against failures and bandwidth fluctuation of the underlying networks. Multimedia traffic can be distributed over those multiple paths in order to maximize paths' utilization and to increase application throughputs. Most of flow-based routing algorithms consider only common metrics such as paths' bandwidth or delay, which may be effective for data applications but not for real-time applications such as Voice over IP (VoIP), in which different levels of such performance metrics may give the same level of the performance experienced by end users. This paper focuses on such VoIP overlay networks and proposes a novel alternative path based flow routing algorithm using an application-specific traffic metric, i.e. "VoIP Path Capacity (VPCap)," to calculate the maximum number of QoS satisfied VoIP flows which may be distributed over each available overlay path at a moment. The simulation results proved that more QoS-satisfied VoIP sessions can be established over the same multiple overlay paths, comparing to traditional approaches.

This paper presents an optimal load balancing algorithm based on both of the ANFIS (Adaptive Neuro-Fuzzy Inference System) modeling and the FIS (Fuzzy Inference System) for the local status of real servers. It also shows the substantial benefits such as the removal of load-scheduling overhead, QoS (Quality of Service) provisioning and providing highly available servers, provided by the suggested method.

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 proposes an adaptive zone selection (AZS) scheme with adaptive modulation for wireless packet transmission systems to achieve high throughput and low delay performances even under non-uniform traffic conditions. In the proposed system, based on the measurement of the propagation path characteristics to each access point (AP) as well as broadcasted blocking probability information from each AP, a terminal autonomously selects an AP and modulation parameters that give the minimum transmission failure probability determined by both the call blocking rate due to lack of radio resource and packet error rate due to severe channel conditions. Computer simulation confirms that the proposed scheme greatly improves throughput and delay performances especially under non-uniform traffic conditions.

This paper describes the effects of traffic distributions on uplink and downlink communications qualities in CDMA cellular systems. Many researches have been done from the viewpoint of the system capacity under ideal conditions in both uplink and downlink. However, there are few studies regarding traffic distributions that concurrently affect the uplink and downlink quality. The characteristics in both links are different even in a spatially uniform traffic distribution because the system structures are not symmetric between both links. When non-uniform radio environments are assumed, both link qualities become very different from each other. It is therefore important to design systems in consideration of link-specific characteristics in whole service area. This paper clarifies the difference in both link characteristics in CDMA systems regarding traffic distributions.

An analytical framework to study the nonuniformity in geographical distribution of the traffic load in low earth orbit satellite communication systems is presented. The model is then used to evaluate the throughput performance of the system with direct-sequence packet spread-slotted ALOHA multiple-access technique. As the result, it is shown that nonuniformity in traffic makes the characteristics of the system significantly different from the results of uniform traffic case and that the performance of each user varies according to its location. Moreover, the interference reached from users of adjacent satellites is shown to be one of the main factors that limit the performance of system.