We analyze the MAC performance of the IEEE 802.11 DCF and 802.11e EDCA in non-saturation condition where device does not have packets to transmit sometimes. We assume that a flow is not generated while the previous flow is in service and the number of packets in a flow is geometrically distributed. In this paper, we take into account the feature of non-saturation condition in standards: possibility of transmission performed without preceding backoff procedure for the first packet arriving at the idle station. Our approach is to model a stochastic behavior of one station as a discrete time Markov chain. We obtain four performance measures: normalized channel throughput, average packet HoL (head of line) delay, expected time to complete transmission of a flow and packet loss probability. Our results can be used for admission control to find the optimal number of stations with some constraints on these measures.

The overall performance of P2P-based file sharing applications is becoming increasingly important. Based on the Adaptive Resource-based Probabilistic Search algorithm (ARPS), which was previously proposed by the authors, a novel probabilistic search algorithm with QoS guarantees is proposed in this letter. The algorithm relies on generating functions to satisfy the user's constraints and to exploit the power-law distribution in the node degree. Simulation results demonstrate that it performs well under various P2P scenarios. The proposed algorithm provides guarantees on the search performance perceived by the user while minimizing the search cost. Furthermore, it allows different QoS levels, resulting in greater flexibility and scalability.

New multimedia services over cellular/WLAN overlay networks require different Quality of Service (QoS) levels. Therefore, an efficient network management system is necessary in order to realize QoS sensitive multimedia services while enhancing network performance. In this paper, we propose a new online network management framework for overlay networks. Our online approach to network management exhibits dynamic adaptability, flexibility, and responsiveness to the traffic conditions in multimedia networks. Simulation results indicate that our proposed framework can strike the appropriate balance between performance criteria under widely varying diverse traffic loads.

To guarantee strict Quality of Service (QoS) for real-time applications, we have previously proposed an output buffer control mechanism in IP routers, confirmed its effectiveness through simulations, and implemented a prototype. This mechanism can guarantee strict QoS within a single router. In this paper, we propose a control scheme of cooperation between IP routers equipped with this mechanism by using one of the signaling protocols. Our proposed scheme aims to stabilize End-to-End (E2E) flow delay within the target delay. In addition, our mechanism dynamically updates reserved resources between IP routers to improve E2E packet loss rate. We present an implemented design of our scheme and an empirical evaluation of the implementation. These results show quantitatively how our scheme improves the quality of video pictures.

Delay jitter degrades the quality of delay-sensitive live media streaming. We investigate the use of multipath transmission with two paths to reduce delay jitter and, in this paper, propose a nearly equal delay path set configuration (NEED-PC) scheme that further improves the performance of the multipath delay jitter reduction method for delay-sensitive live media streaming. The NEED-PC scheme configures a pair of a maximally node-disjoint paths that have nearly equal path delays and satisfy a given delay constraint. The results of our simulation experiments show that path sets configured by the NEED-PC scheme exhibit better delay jitter reduction characteristics than a conventional scheme that chooses the shortest path as the primary path. We evaluate the performance of path sets configured by the NEED-PC scheme and find that the NEED-PC scheme reduces delay jitter when it is applied to a multipath delay jitter reduction method. We also investigate the trade-off between reduced delay jitter and the increased traffic load incurred by applying multipath transmission to more flows. The results show that the NEED-PC scheme is practically effective even if the amount of additional redundant traffic caused by using multipath transmission is taken into account.

With the rapid advances in wireless network communication, multimedia presentation has become more applicable. However, due to the limited wireless network resource and the mobility of Mobile Host (MH), QoS for wireless streaming is much more difficult to maintain. How to decrease Call Dropping Probability (CDP) in multimedia traffic while still keeping acceptable Call Block Probability (CBP) without sacrificing QoS has become an significant issue in providing wireless streaming services. In this paper, we propose a novel Dynamic Resources Adjustment (DRA) algorithm, which can dynamically borrow idle reserved resources in the serving cell or the target cell for handoffing MHs to compensate the shortage of bandwidth in media streaming. The experimental simulation results show that compared with traditional No Reservation (NR), and Resource Reservation in the six neighboring cells (RR-nb), and Resource Reservation in the target cell(RR-t), our proposed DRA algorithm can fully utilize unused reserved resources to effectively decrease the CDP while still keeping acceptable CBP with high bandwidth utilization.

We present a simple proactive time rearrangement scheme (PATRA) that reduces the interferences from multi-radio devices equipped in one platform and guarantees user-conceived QoS. Simulation results show that the interference among multiple radios in one platform causes severe performance degradation and cannot guarantee the user requested QoS. However, the PATRA can dramatically improve not only the user-conceived QoS but also the overall network throughput.

Energy-efficient transmission scheme is very essential for Wireless Personal Area Networks (WPNs) for maximizing the lifetime of energy-constrained wireless devices and assuring the required QoS in the actual physical transmission at each allocated TDMA time slot. We therefore propose the minimum energy (ME) criterion based adaptive transmission scheme which determines the optimum combination of transmit power, physical data rate and fragment size required to simultaneously minimize the energy consumption and satisfy the required QoS in each assigned time duration. The improved performances offered by the proposed algorithm are demonstrated via computer simulation in terms of throughput and energy consumption.

Rapid advances in wireless sensor networks require routing protocols which can accommodate new types of power source and data of differing priorities. We describe a QoS-aware geographic routing scheme based on a solar-cell energy model. It exploits an algorithm (APOLLO) that periodically and locally determines the topological knowledge range (KR) of each node, based on an estimated energy budget for the following period which includes the current energy, the predicted energy consumption, and the energy expected from the solar cell. A second algorithm (PISA) runs on each node and uses its knowledge range to determine a route which meets the objectives of each priority level in terms of path delay, energy consumption and reliability. These algorithms maximize scalability and minimize memory requirements by employing a localized routing method which only uses geographic information about the host node and its adjacent neighbors. Simulation results confirm that APOLLO can determine an appropriate KR for each node and that PISA can meet the objectives of each priority level effectively.

Wireless content sharing where peers share content and services via wireless access networks requires user contributions, as in fixed P2P content sharing. However, in wireless access environments, since the resources of mobile terminals are strictly limited, mobile users are not as likely to contribute as ones in fixed environments. Therefore, incentives to encourage user contributions are more significant in wireless access environments. Although an incentive service differentiation architecture where the content transfer rate is adjusted according to the contributions of each downloading user has been already proposed for fixed P2P, it may not work well in wireless access environments because several factors effect wireless throughput. In this paper, we propose a novel architecture for contribution-based transfer-rate differentiation using wireless quality of service (QoS) techniques that motivates users to contribute their resources for wireless content sharing. We also propose a radio resource assignment method for our architecture. Computer simulations and game-theoretic calculations validate our architecture.

Wireless mesh networks are attracting more and more attention as a promising technology for the next generation access infrastructure. QoS support is a unavoidable task given the rising popularity of multimedia applications, and also a challenging task for multi-hop wireless mesh networks. Among the numerous QoS factors, end-to-end delay is one of the most critical and important issues, especially for the real time applications. Over multi-hop wireless mesh networks, end-to-end delay of a flow is highly dependant on the number of hops as well as congestion condition of the hop nodes that the flow traverses through. In this paper, we propose QoS priority control schemes based on the end-to-end QoS delay metrics in order to increase traffic accommodation, i.e., the numbers of real-time flows which satisfy the requirements of end-to-end delay and packet delivery ratio over multi-hop wireless mesh networks. The first scheme enables source and forwarding nodes to perform priority control based on the number of hops of routes. The second scheme enables nodes to perform priority control based on the congestion condition of the hop nodes, where the flow traverses through. The effectiveness of the proposed schemes is investigated with NS-2 network simulator for voice and video traffics over multi-hop wireless mesh networks. Simulation results show that the scheme greatly improves the traffic accommodation for voice and video applications in multi-hop wireless mesh networks.

In this paper, we propose the use of a discrete-time connection oriented Internet service system with a release delay for broadband, high-speed, high-capacity and high-reliability Internet requirements. The release delay called close-delay is set before the release process of a connection. An upper limit length T called timer length is set as a system parameter for the close-delay period. We build a batch arrival Geom*/G/1 queue model with a setup/close-delay/close-down strategy to characterize the system operation. By using a discrete-time imbedded Markov chain approach, we derive the stationary distribution of the system, and present the formula for Probability Generation Functions of the queue length, waiting time, busy period and busy cycle. Correspondingly, we describe the performance measures for the packet response time, setup ratio, and utility of connection. We also develop a cost model to determine the optimal timer length and its expected optimal cost. Based on numerical results, we discuss the influence of the timer length for the close-delay period on the system performance and investigate the minimum timer length and the minimum cost for different offered loads and different burst degrees, and show that the choice of the timer length is significant in improving the system performance.

Quality of service (QoS) control is one of the key technologies for the next-generation network (NGN). In the conventional method, the bearer quality on the carrier network is managed, but the end-to-end QoS for end users needs to be guaranteed. The quality of a public switched telephone network (PSTN) is guaranteed, and the quality of the telephone terminal is also stable. Therefore, end-to-end quality management of PSTN services has been achieved. However, the quality of neither IP networks nor VoIP services is guaranteed in general. In addition, there are numerous VoIP terminals and the differences in their implementations strongly affect the speech quality experienced by end users. Thus, quality management technologies need to be embedded in the VoIP terminal to achieve the equivalent of end-to-end QoS management for the PSTN. These technologies are recommended in IETF RFC3611 "RTCP XR" and ITU-T recommendation P.564 as the framework for end-to-end quality management, but their usage is not shown. Therefore, we propose an end-to-end quality management method for VoIP speech using RTCP XR. We realize an end-to-end QoS monitoring method between the customer and the operator of the carrier network. We define the parameters that should be implemented in RTCP XR packets to estimate the quality of VoIP services based on experimental results.

In order to find paths guaranteed by Quality of Service (QoS), the link state database (LSDB), containing QoS constraint information, and residing in routers, needs to be well managed. However, there is a trade-off between the exact reflection of the current link status and the update cost to calculate and maintain this data. In order to perfectly reflect the current link state, each router immediately notifies its neighbors whenever link state information changes. However, this may degrade the performance of the router. On the other hand, if current link state information is not updated routinely, route setup requests may be rejected because of the discrepancy between the current link state information and the previously updated link state information in the LSDB. Therefore, we need link state update (LSU) algorithms making it possible to appropriately update the LSDB. In addition, to facilitate implementation, they also should have low-complexity and must be adaptive under the variation of network conditions. In this paper, we propose an enhanced simple-adaptive (ESA) LSU algorithm, to reduce the generation of LSU messages while maintaining simplicity and adaptivity. The performance of this algorithm is compared with five existing algorithms by rigorous simulations. The comparision shows that the ESU algorithm can adapt to changes in network conditions and its performance is superior to existing LSU algorithms.

This paper describes IP encapsulation technologies for the Mobile RSVP tunnel in next generation networks. Bandwidth is inherently a scarce network resource, and hence signaling overhead should be minimized as much as possible. However, because of duplicate RSVP messages, the existing RSVP tunnel-based mechanism suffers from bandwidth overhead and tunnel problems. The waste of network resources prevents low-cost network construction and the maximization of integrated network utility, which are the goals of next generation networks, and can lower the reliability of networks with the increase of service subscribers and resultant expansion of resource consumption. To solve these problems and to support end-to-end QoS efficiently, RSVP needs to be changed at a minimum degree. In this paper, a new IP encapsulation mechanism for saving of network resources in the Mobile RSVP tunnel (IPEnc-RSVP) is proposed. In order to compare the proposed mechanism and the existing RSVP tunnel-based mechanism in Mobile IP-based networks, we perform a comparative analysis of bandwidth consumption gain, throughput, mean packet delay, etc., and demonstrate the superiority of the proposed mechanism. In addition, we analyze several performance factors of RSVP protocols by applying the existing RSVP tunnel-based mechanism and the proposed mechanism, respectively.

In this letter, we propose PQ-MAC: a priority-based medium access control (MAC) protocol for providing quality of service (QoS) in wireless sensor networks (WSNs) which minimizes the energy consumption with traffic-based sleep-wakeup scheduling and supports QoS using differentiating channel access policy, packet scheduling, and queue management. The PQ-MAC utilizes the advantages of time division multiple access (TDMA) and slotted carrier-sense multiple access (CSMA). The proposed protocol is an energy-efficient, priority-based, and QoS compatible MAC protocol. It consists of multi-level queue management, sleep-wakeup scheduling, and an ordered contention period (CP) scheme. It also guarantees time-bounded delivery of QoS packets. Performance evaluation is conducted between PQ-MAC and S-MAC with respect to three performance metrics: energy consumption, throughput, and average latency. The simulation results show that the performance of PQ-MAC is better than that of S-MAC.

Future optical code division multiple access (CDMA) networks should be designed for multirate and fully integrated multimedia services. In the conventional schemes, multilength optical orthogonal codes (OOCs) are designed to support multirate systems, while variable-weight OOCs are designed to support differentiated quality of service (QoS) for multimedia applications. In this paper, a novel class of optical signature codes; multiple-length variable-weight optical orthogonal codes (MLVW-OOC) is proposed for supporting multirate and integrated multimedia services in optical CDMA networks. The proposed MLVW-OOC has features that are easy to construct variable-weight codes and expanded to multiple-length codes. A construction method for designing MLVW-OOCs up to three levels of codes is discussed. The designed MLVW-OOCs can support differentiated requirements on data rates and QoS for several types of services in the networks. A code analysis for obtaining the value of cross-correlation constraints or multiple access interference (MAI) computation for several levels of codes is also suggested. The cross-correlation constraints of the proposed codes are better than the conventional codes such as multilength OOCs. Finally, the bit error probability performance of the two-level MLVW-OOC is evaluated analytically. The results show that the proposed MLVW-OOC can provide differentiated bit error probability performances for several combinations of data rates and QoS.

In real-time systems, deadline misses of the tasks cause a degradation in the quality of their results. To improve the quality, we have to allocate CPU utilization for each task adaptively. Recently, Buttazzo et al. address a feedback scheduling algorithm, which dynamically adjusts task periods based on the current workloads by applying a linear elastic task model. In their model, the utilization allocated to each task is treated as the length of a linear spring and its flexibility is described by a constant elastic coefficient. In this paper, we first consider a nonlinear elastic task model, where the elastic coefficient depends on the utilization allocated to the task. We propose a simple iterative method for calculating the desired allocated resource and derive a sufficient condition for the convergence of the method. Next, we apply the nonlinear elastic model to an adaptive fair sharing controller. Finally, we show the effectiveness of the proposed method by computer simulation.

An adaptive tri-threshold dynamic call admission control scheme for wideband mobile cellular networks is proposed. The relationship between the Channel Utilization and the Weighted Handover Dropping Probability versus traffic loadings are investigated. This scheme supports voice, data and multimedia services with differentiated QoS.

Interactive audio-video applications over IP networks have subjective tradeoffs between fidelity and latency owing to packet buffering at the receiver. Increasing the buffering time improves the fidelity, whereas it degrades the latency. This paper makes the subjective tradeoff between fidelity and latency clear in a quantitative way. In addition, we examine the effect of tasks on the subjective tradeoff. In evaluating the effect of tasks, we use two tasks according to ITU-T Recommendation P.920. An experiment was conducted to measure user-level QoS of an interactive application with the psychometric methods. We then investigate the subjective tradeoff quantitatively by QoS mapping. The experimental results confirm that there exists the buffering time which makes user-level QoS the highest. The results also show that the optimum buffering time depends on the kind of task.