This paper proposes a distributed adaptive routing algorithm that may be applied to inter-domain calls passing over any type of network topology, traffic management and switching techniques on the path, while carrying bidirectional traffic with multiple QoS requirements. The path is searched within a contour area restricted by the number of hops between source and destination while the end-to-end admission of calls is controlled at source node and each hop's admission at each node, reflecting the latest resources availability and network conditions for the given QoS requirements. Performance analysis in various conditions shows good applicability in real networks.

This paper considers an efficient scheduling policy for Bluetooth Medium Access Control (MAC) and its parameter optimization method. The proposed algorithm improves performance as well as supports Quality of Service (QoS) simultaneously. Since Bluetooth is basically operated with a Round Robin (RR) scheduling policy, many slots may be wasted by POLL or NULL packets when there is no data waiting for transmission in the queues of the polled pair. To overcome this link wastage problem, several algorithms have been proposed. However, they have some limitations such as a heavy signaling overhead or no consideration of QoS. Therefore, we have proposed an efficient Bluetooth MAC scheduling algorithm, Differentiated K-Fairness Policy (Diff-KFP), which guarantees improved throughput and delay performance, and it can also lead to differentiated services. That is, if the parameter of the proposed algorithm is optimized, we can satisfy the QoS requirement of each master-slave pair and thereby keep communications in progress from interruption, which is a source of throughput degradation. Simulation results show that our algorithm has remarkably improved the performance and gratifies the QoS requirements of various applications.

This paper proposes a unified packet scheduling method that considers the delay requirement of each traffic data packet whether real time (RT) or non-real time (NRT), the channel conditions of each accessing user, and the packet type in hybrid automatic repeat request (ARQ), i.e., either initially transmitted packet or retransmitted packet, in the forward link for Orthogonal Frequency and Code Division Multiplexing (OFCDM) wireless access. In the proposed packet scheduling method, the overall priority function is decided based on PTotal = αDelayPDelay + αTypePType + αSINRPSINR (PDelay, PType, and PSINR are the priority functions derived from the delay requirement, type of packet, and the received signal-to-interference plus noise power ratio (SINR), respectively, and αDelay, αType, and αSINR are the corresponding weighting factors). The computer simulation results show that the weighting factor of each priority function as αType/αDelay = 0.6, αSINR/αDelay = 0.4 assuming the linear-type function in PDelay and a constant-type function in PType is optimized. Furthermore, we show that the outage probability for achieving the packet loss rate (PLR) of less than 10-3 for non-real time (NRT) traffic users employing the proposed packet scheduling method is reduced by approximately two orders of magnitude compared to that using the Priority Queuing (PQ) method while maintaining the PLR of real-time (RT) traffic users at the same level as that using the PQ method.

In this study, we propose an object-based multimedia model for specifying the QoS (quality of service) requirements, such as the maximum data-dropping rate or the maximum data-delay rate. We also present a resource allocation model, called the net-profit model, in which the satisfaction of user's QoS requirements is measured by the benefit earned by the system. Based on the net-profit model, the system is rewarded if it can allocate enough resources to a multimedia delivery request and fulfill the QoS requirements specified by the user. At the same time, the system is penalized if it cannot allocate enough resources to a multimedia delivery request. We first investigate the problem of how to allocate resources efficiently, so that the QoS satisfaction is maximized. However, the net-profit may be distributed unevenly among the multimedia delivery requests. Thus, the second problem discusses how to allocate the resource efficiently so that the net-profit difference is minimized between any two multimedia requests. A dynamic programming based algorithm is proposed to find such an optimal solution with the minimum net-profit differences.

In this paper, we propose an autonomously distributed QoS control method for MPEG video streaming in a wide area network. The capacity of the links and the characteristics of video streams change dynamically. However, managing the condition of all the links and streams in the network is difficult. In the proposed method, the routers in the network monitor the conditions of the links and streams locally and control the transmission rate of the stream server. Picture-quality oriented fairness is achieved by reducing the transmission rate of the streams with the higher PSNR in the bottleneck link. The computer simulation results show that the proposed method can be applied to a wide area network.

AF service class in Diffserv by realizes minimum bandwidth guarantee with the use of differentiated drop precedence property marked on each packet. In the context of a multiple domains environment, however, QoS of individual flow is not always preserved due to the re-marking behavior forced at the domain boundaries. Focusing on this point, this paper proposes new packet re-marking schemes that can improve the per-flow QoS of AF service traversing multiple domains. The basic concept of the schemes distinguishes packets re-marked to out-of-profile at the domain boundaries from those already marked as out-of-profile at the time of entering the network, and allows the re-marked packet to recover back to in-of-profile, thus regaining its rightful QoS within the networks. The performance of the proposed schemes is evaluated through simulation. The results on UDP flows show the effectiveness of the proposed schemes for reducing packet losses on the flow through multiple domains and preserving fairness between flows. Simulations on TCP flows show that the proposed schemes improve the throughput of the flows through multiple domains. The proposed scheme is especially effective on the transfer time of short TCP flows such as Web traffic, whose throughput is affected more seriously by a single packet loss due to its flow control mechanisms.

This paper presents an application-level QoS comparison of three inter-destination synchronization schemes: the master-slave destination scheme, the synchronization maestro scheme, and the distributed control scheme. The inter-destination synchronization adjusts the output timing among destinations in a multicast group for live audio and video streaming over the Internet/intranets. We compare the application-level QoS of these schemes by simulation with the Tiers model, which is a sophisticated network topology model and reflects hierarchical structure of the Internet. The comparison clarifies their features and finds the best scheme in the environment. The simulation result shows that the distributed control scheme provides the highest quality of inter-destination synchronization among the three schemes in heavily loaded networks, while in lightly loaded networks the other schemes can have almost the same quality as that of the distributed control scheme.

This paper proposes a new FEC scheme using backup channel to send redundant information instead of piggybacking the main packet. This is particularly applicable to the modern IP networks which are distributed all over the world. In this method only one source coder for both the main and the redundant payload is used to reduce the overall computational complexity. The Gilbert loss model (GLM) is employed to verify the improvement of the packet loss probability in this new method compared with that in a single path FEC scheme. It is shown, through simulation results that using our proposed backup channel can considerably improve the packet loss and delay performance of the VoIP networks.

Proposed CCRSVP (CandidateCasting RSVP) algorithm is a new fast handoff method for IEEE 802.11 Wireless LAN (WLAN) environments. It shows good performance in the handoff latency and the bandwidth efficiency aspect and guarantees QoS because it uses an advanced multicasting method and RSVP. CCRSVP uses L2 information (BSSID) of WLAN and starts reserving resources and multicasting packets before L2 handoff completes. Therefore, the proposed algorithm can reduce L3 handoff latency more than other methods. To show performance of CCRSVP algorithm, we calculate handoff latency and packet loss ratio of each algorithm. Also we model handoff process which uses RSVP mechanism to confirm resource efficiency. Proposed handoff model uses parameters which can distinguish each handoff algorithm. We introduce Markov chain which can analyze handoff model and analysis method which uses iteration method. In this article, the results show that the proposed algorithm shows superior bandwidth efficiency than existing L3 handoff algorithms using RSVP. To analyze bandwidth efficiency of each algorithm, we compare the blocking probability which occurs in case of absence of resource, resource usage which shows reservation quantity, the average number of ongoing session which really uses resource reserved and resource utilization. We can confirm that CCRSVP algorithm has better performance than other algorithms at each comparative item.

This paper proposes dynamic distributed unicast and multicast routing algorithms for multiple classes of QoS guaranteed networks. Each link in such a network is assumed to be able to provide multiple classes of QoS guarantee by reserving various amounts of resource. A distributed unicast routing algorithm, DCSP (Distributed Constrained Shortest Path), for finding a QoS constrained least cost path between each O-D (Originating-Destination) pair, is proposed first. Two class reduction schemes, the linear and logarithmic policies, are develpoed to prevent exponential growth of the number of end-to-end QoS classes. Based on DCSP, two distributed multicast routing algorithms, DCSPT (Distributed Constrained Shortest Path Tree) and DTM (Distributed Takahashi and Mutsuyama), are proposed to find QoS constrained minimum cost trees. Numerical results indicate that DCSP strongly outperforms previously proposed centralized algorithms and it works better with the linear class reduction method. For the multicast routing algorithms, the DCSPT with linear class reduction method yields the best performance of all multicast routing algorithms.

In this paper, we propose the Synchronized Mobile Multicast (SMM) scheme for the real-time multimedia service to achieve three most important characteristics that the traditional Home Subscription (HS) and Remote Subscription (RS) mobile schemes cannot support. First, the SMM scheme supports the scalable one-to-many and many-to-many synchronized multimedia multicast on mobile IP networks to achieves seamless playback of continuous media streams even when both the mobile sender and receivers handoff simultaneously. Second, it analyzes the minimal buffer requirements of the mobile sender, the core router, the foreign agents and the mobile receivers in the multicast tree and formulates the initial playback delay within a handoff Guarantee Region (GR). Further, combined with the fine granularity scalability (FGS) encoding approach in the MPEG-4 standard, the SMM scheme achieves superior multimedia QoS guarantees and unlimited numbers of handoffs of the mobile sender and receivers only at the cost of degraded video quality for a short period after handoff with minimal extra bandwidth.

In future high-speed networks, provision of diverse multimedia services with strict quality-of-service (QoS) requirements, such as bandwidth, delay and so on, is desired. QoS routing is a possible solution to handle these services. Generally, a path selection for QoS routing is formulated as a shortest path problem subject to multiple constraints. However, it is known to be NP-complete when more than one QoS constraint is imposed. As a result, many heuristic algorithms have been proposed so far. The authors proposed a path selection algorithm Fallback+ for QoS routing, which focuses not only on the path selection with multiple constraints but also on the efficient use of network resources. This paper proposes an enhanced version of Fallback+, named Enhanced Fallback+, where in a shrewd way, it keeps tentative paths produced in the conventional Fallback algorithm with Dijkstra's algorithm. Simulation experiments prove the excellent performance of Enhanced Fallback+, compared with the original Fallback+ and other existing path selection algorithms.

We study the statistical multiplexing performance of self-similar traffic. We consider that input streams have different QoS (Quality of Service) requirements such as loss and delay jitter. By applying the FBM (fractal Brownian motion) model, we present methods of estimating the effective bandwidth of aggregated traffic. We performed simulations to evaluate the QoS performances and the bandwidths required to satisfy them. The comparison between the estimation and the simulation confirms that the estimation could give rough data of the effective bandwidth. Finally, we analyze the bandwidth gain with priority multiplexing against non-prioritized multiplexing and suggest how to get better performance with the right configuration of QoS parameters.

The issue of guaranteeing Quality of Services (QoS) in a network has emerged in recent years. The Proportional Delay Differentiated Model has been presented to provide the predictable and controllable queueing delay differentiation for different classes of connections. However, most related works have focused on providing this model for a wired network. This study proposes a novel scheduler to provide proportional delay differentiation in a wireless network that includes a multi-state link. This scheduler, Look-ahead Waiting-Time Priority (LWTP), offers proportional delay differentiation and a low queueing delay, by adapting to the location-dependent capacity of the wireless link and solving the head-of-line (HOL) blocking problem. The simulation results demonstrate that the LWTP scheduler actually achieves delay ratios much closer to the target delay proportion between classes and yields smaller queueing delays than past schedulers.

This paper examines the effect of segmentation mismatch on audio-video transmission by Bluetooth. We focus on the segmentation mismatch caused by the difference between the RFCOMM Maximum Frame Size and the baseband packet payload size. By experiment, we assessed the maximum throughput and media synchronization quality for various types of ACL packets. In the experiment, a media server transferred stored video and audio streams to a single terminal with point-to-point communication; we supposed no fading environment and added white noise by which interference from DSSS systems is modeled. The experiment showed that the effect of segmentation mismatch is large especially when the total bit rate of the two streams is near the channel transmission rate. We also observed that the media synchronization control is effective in compensating for the disturbance by the segmentation mismatch in noisy environments.

In this paper, the design of a QoS scheduling scheme for the Internet traffic is proposed by considering the correlation property of the arriving traffic. The basic concept of the Weighted Fair Queuing (WFQ) is adopted in the proposed scheme, however, the correlation property of the traffic stream is applied as the heuristic to adjust the share weight factors of each traffic type dynamically. The Auto Regressive Integrated Moving Average (ARIMA) model is applied in this paper to characterize the correlation property of the Internet traffic. And the share weight factors are derived from the parameters of the AR part and MA part. Experimental simulations are performed to illustrate the effectiveness of the proposed scheme. In addition to comparing the performance of each service types, we also define a fair play parameter (FPP) to examine the fairness index among various traffic streams of the proposed scheme. The experimental results indicate that the proposed scheme demonstrates a quite good performance in scheduling the integrated services and the fairness among service classes can also be achieved, especially when the link capacity is limited.

Differentiated Services architecture provides a framework that enables relative differentiation of Assured Forwarding (AF) service. The differentiation is quantified by QoS parameters in terms of loss probability and maximum delay. We develop herein an efficient model to compute resource allocation in terms of buffer and service rate that satisfies the QoS differentiation between classes of service. To evaluate the performance of the proposed model, we conducted extensive simulation on both single-node and multi-node cases. The simulation studies show that the model can provide an efficient method to allocate network resources for aggregated traffic.

This paper investigates the relations between IP network performances and the speech quality of the Voice over IP (VoIP) service through extensive experiments on a test bed network. The aim is to establish an effective and practical methodology for telecommunications operators to manage the quality of VoIP service via the management of IP network performances under their control. As IP network performances, utilization of the bottleneck link in the test bed and the following statistical factors of VoIP packets are examined: the standard deviation of delay variations (jitters), the standard deviation of packet interarrival times, and the packet loss ratio. On the other hand, VoIP speech quality is monitored as the Perceptual Evaluation of Speech Quality (PESQ). To investigate the relations under various network conditions, the experiments are performed by varying the following network related parameters of the test bed: the bandwidth of the bottleneck link, the size of the bottleneck buffer, the propagation delay, and the average of the data sizes transmitted as background data traffic. Statistical analyses of the experimental results suggest that managing the standard deviation of jitters in a network serves as a promising methodology, because its close relation to VoIP speech quality possesses robustness to changes in the network conditions. The robustness makes it practically useful since telecommunications operators can apply it to their networks, which are subject to change. The findings in this paper have opened up new visions for telecommunications operators to manage the Quality of Service (QoS) of VoIP service.

VoIP is one of the key technologies for recent telecommunication services. The quality of its services should be discussed in subjective terms. Since subjective quality assessment is time-consuming and expensive, however, objective quality assessment which estimates subjective quality without carrying out subjective quality experiments is desirable. This paper discusses the performance of the objective quality measure that was standardized as ITU-T Recommendation P.862 and clarifies the quality factors that can be evaluated with satisfactory accuracy based on it. We found that P.862 can be applied to the evaluation of coding distortion, tandeming of codecs, transmission bit-errors, packet loss, and silence compression in a codec, at least for clean Japanese speech. In addition, we propose a method of estimating the subjective quality evaluation value from objective measurement results and show the validity of this method. We also evaluate the uniqueness of objective quality assessment based on P.862 from the viewpoints of the effect of measurement noise and the variation of test speech samples, and propose how to improve the reproducibility of objective quality assessment.

This paper proposes a novel method of identifying the design parameters for a practical implementation of the fair queueing discipline, which is capable of class-level delay control. The notion of class weight is introduced at first, and then the session weights are determined. This two-phase approach is favorable in terms of the scalability;that is, the overall complexity is dependent upon the number of classes only. We propose a packet scheduler referred to as the DPS (Delay-centric Processor Sharing) scheme which employs those design parameters to deliver class-wise delay bound services. The associated admission policy for delay guarantee is also derived. System analysis and derivation of the parameters have their origins in the understanding of the so-called system equation, which describes the dynamics of the class-level service share. The proposed design parameters are QoS-aware in that they are consistently refined depending on the system status. Several numerical and simulation results show that the DPS scheme is advantageous over other ones in terms of both resource efficiency and the robustness. Concerning the scalability, we show that an alternative tagging process of the DPS scheme is implementable with O(1) complexity with no significant degradation in delay performance.