There is a growing demand that mobile networks should provide quality-of-service (QoS) to mobile users since portable devices become popular and more and more applications require real-time services. Providing QoS to mobile hosts is very difficult due to mobility of hosts. The resource ReSerVation Protocol (RSVP) establishes and maintains a reservation state to ensure a given QoS level between the sender and receiver. However, RSVP is designed for fixed networks and thus it is inadequate in wireless mobile networking environments. In this paper, we propose a resource reservation protocol for mobile hosts in mobile networks. The proposed protocol extends the RSVP by introducing RSVP agents in local networks to manage the reservations. The proposed protocol reduces packet delay, bandwidth overhead, and the number of RSVP messages to maintain reservation states. We examined the performance of the proposed protocol by simulation and we got an improved performance over the existing protocols.

Appropriate test signals defined by formula or generated by algorithm are used for measuring objective QoS (Quality of Services) for voice operated telecommunication devices such as telephone and speech codec (coder-decoder). However, that for measuring residual echo characteristics in hands-free telecommunications equipped with acoustic echo canceller is under study in ITU-T Recommendation G.167. This paper describes comparative assessment of test signals for measurement of residual echo characteristics. In hands-free telecommunications, acoustical echo canceller has been developed to remove a room echo signal through the loudspeaker to the microphone in the receiving end. Performance of the echo canceller system is evaluated by residual echo characteristics expressed in echo return loss enhancement (ERLE). The ERLE can be conventionally measured by putting white noise into the echo canceller system. However, white noise is not adequate as the test signal for measuring the performance of the echo canceller, since the performance may depend on the characteristics of input test signal, and the characteristics of the white noise differ from those of real voice. Therefore, this paper discusses appropriate characteristics of real voice required for objective quality evaluation of echo canceller system. The test signals used for this verification tests were real voice (RV), white noise (WN), frequency weighted noise (FWN), artificial voice (AV), and composite source signal (CSS) depending on the approximation of real voice characteristics. As the comparative assessment results, the ERLE characteristics measured by artificial voice conforming to ITU-T Recommendation P.50 having average characteristics of real voices in time and frequency domains are almost equivalent to those of real voice and best among those test signals. It is concluded that artificial voice P.50 is satisfied with measurement of residual echo characteristics.

With the aim of enabling concatenative synthesis of expressive speech, we herein report progress towards developing robust and automatic algorithms for paralinguistic annotation of very large recorded-speech corpora. In particular, we describe a method of combining robust acoustic-prosodic and cepstral analyses to locate centres of acoustic-phonetic reliability in the speech stream, wherein physiologically meaningful parameters related to voice quality can be estimated more reliably. We then report some evaluations of a specific voice-quality parameter known as the glottal Amplitude Quotient (AQ), which was proposed in [2],[6] and is here measured automatically at centres of reliability in continuous speech. Analyses of a large, single-speaker corpus of emotional speech first validate the perceptual importance of the AQ parameter in quantifying the mode of phonation along the pressed-modal-breathy continuum, then reveal some of its phonetic, prosodic, and paralinguistic dependencies.

As multimedia and high-speed traffic become more popular on the Internet, the various traffic requiring different qualities of service (QoS) must co-exist. In addition, classified services based on Diff-Serv (Differentiated Service), MPLS (Multi-Protocol Label Switching), etc., have come into wide use. Today's Internet environment requires routers to perform control mechanisms in order to guarantee various QoSs. In this paper, we propose a smart buffer management scheme for the Internet router that uses hierarchical priority control with port class and flow level. Furthermore, since the proposed scheme must operate at very high speed, we first propose several design policy for high speed operation and the hardware implementation is performed in VHDL code. Implementation results show that the proposed scheme can scale with high-speed link, achieving the maximum rate of 4.0 Gbps by using the 3.5 µm CMOS technology.

Class-based delay differentiation model has been recently proposed as a part of relative differentiated services frameworks, and it is shown that the model can provide delay differentiation without admission control and end-to-end resource reservation. In this paper, however, we observe that there can be inconsistent delay differentiation caused by different size of packets. We propose packet size-based delay differentiation model and show that packet size-based queueing is effective to achieve equal delay within a class and provide consistent delay differentiation between classes through simulations. Simulation results also show that the proposed model improves jitter characteristics of CBR flows.

To support video streaming over the current Internet, a media server usually needs to perform adaptive streaming in combination with congestion control. While existing streaming schemes are mainly designed for particular congestion algorithms and rate shaping techniques, this paper proposes a general buffer-driven adaptive streaming scheme based on control theory. Our scheme can be applied in combination with different congestion control algorithms and different rate shaping (and source coding) techniques. It is also amenable to analysis and easy to implement. Analytical and simulation results show that the proposed scheme gives satisfactory video quality under various Internet conditions while fully utilizing the available network bandwidth.

In this paper, we propose an efficient quality-providing scheme to satisfy delay bound and loss ratio requirements for real-time video applications. To utilize network resources more efficiently while meeting service requirements, the network resources are dynamically allocated to each video connection based on the predicted traffic and currently provided quality of service degree. With the proposed bandwidth allocation method, a fair quality of service support in terms of packet loss ratio and maximum packet transfer delay to each video source can be achieved. To avoid possible quality violation by incoming new video connections, we present a connection admission control based on the provided QoS for existing connections and the measured traffic statistics. Simulation results show that our proposed dynamic method is able to provide accurate quality control.

This paper presents a novel system-level design methodology, called quality-driven design, by which application-specific optimization can be achieved; furthermore the entire functionality can be shared to maximize design reuse. As a case of study, this paper focuses on quality-driven design for video applications and introduces an output quality adaptive approach based on variable bitwidth optimization to explore a new design space. MPEG2 video is used as the driver application to illustrate the potential of the presented methodology. Experimental results show the effectiveness of the methodology.

In this paper, we present the following schemes for a content-based image search: (1) A fast image search algorithm that can significantly reduce similarity calculation compared to a full comparison of every database image. (2) A compact image representation scheme that can describe the global/local information of the images and provide successful retrieval performance. For fast searches, a tree is constructed by successfully dividing nodes into the desired depth level by working from the root to the leaf nodes using the k-means algorithm. When the query is completed, we traverse the tree top-down by minimizing the route taken between the query image and node centroid until we meet the undivided nodes. Within undivided nodes, the algorithm of triangle inequality is used to find the images most similar to the query. For compact image representation, RGB color histogram features which are quantized into 16 bins each of the R, G, and B channels are used for global information. Dominant hue, saturation, and value which are extracted from the HSV joint histogram in the localized regions within the image are used for local information. These features are sufficiently compact to index image features in large database systems. For experiments on the retrieval efficiency, the use of the proposed method provided substantial performance benefits by reducing the image similarity calculation up to an average of a 96% and for experiments on the retrieval effectiveness, in the best case, it provide a 36.8% recall rate for a whale query image and a 100% precision rate for an eagle query image. The overall performance was a 20.0% recall rate and a 72.5% precision rate.

In this paper, we present new stability conditions for a class of large-scale hybrid dynamical systems composed of a number of interconnected hybrid subsystems. The stability conditions are given in terms of discontinuous Lyapunov functions of the stable hybrid subsystems. Furthermore, the stability conditions are represented by LMIs (Linear Matrix Inequalities) which are computationally tractable.

This letter addresses a neural network (NN)-based predictor for the LP (Linear Prediction) residual. A new NN predictor takes into consideration not only prediction error but also quantization effects. To increase robustness against the quantization noise of the nonlinear prediction residual, a constrained back propagation learning algorithm, which satisfies a Kuhn-Tucker inequality condition is proposed. Preliminary results indicate that the prediction gain of the proposed NN predictor was not seriously decreased even when the constrained optimization algorithm was employed.

This paper presents the performance modeling, analysis, and simulation of SIP-T (Session Initiation Protocol for Telephones) signaling system in carrier class packet telephony network for NGN (Next Generation Networks). Until recently, fone of the greatest challenges in the migration from existing PSTN (Public Switched Telephone Network) toward NGN is to build a carrier class packet telephony network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new packet telephony technology. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) draft is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of PSTN with carrier class packet telephony network. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (Media Gateway Controller) to set up, tear down, and manage VoIP (Voice over IP) calls in carrier class packet telephony network. In this paper, we derive the buffer size, the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (Quality of Service) and system performance of MGC in carrier class packet telephony network focused on toll by-pass or tandem by-pass of PSTN. First, we assume a mathematical model of the M/G/1 queue with non-preemptive priority assignment to represent SIP-T signaling system. Second, we derive the formulas of buffer size, queueing delay, and delay variation for the non-preemptive priority queue by queueing theory respectively. Besides, some numerical examples of buffer size, queueing delay, and delay variation are presented as well. Finally, the theoretical estimates are shown to be in excellent consistence with simulation results.

Future wireless communication systems will provide mobile terminals with high bit-rate transmissions for accessing broadband wired networks. In this paper, we envisage a Time Division Multiple Access - Time Division Duplexing (TDMA-TDD) air interface and we propose a Medium Access Control (MAC) protocol, named Dynamic Scheduling - TDD (DS-TDD), that allows guaranteeing the QoS of different traffic classes and efficiently supports uplink/downlink traffic asymmetries. The DS-TDD performance is theoretically analyzed. Moreover, the DS-TDD protocol is compared with another scheme proposed in the literature. Finally, the impact of packet errors on the DS-TDD performance is evaluated.

In this paper, we propose a new fair queueing algorithm to improve cell delay variation (CDV) for real-time service categories and to make efficient use of system resources for multimedia traffic in high speed ATM networks. The proposed algorithm is called the delay variation-based fair queueing (DVFQ) algorithm, which is based on per-VC queueing to improve CDV and fairness for each VC of real-time services such as CBR and rt-VBR. In DVFQ algorithm, we define two fairness indexes, which indicate the degree of the fairness of CDV at the rate of each VC, and the degree of impartially sharing the bandwidth between the scheduled cells for each VC. The simulation results for both heavily and lightly loaded conditions show that DVFQ algorithm provides better performances in terms of the CDV, the CDV fairness, and the service fairness than those of FCFS for real-time service.

This paper presents an efficient scheme for access bandwidth control for VBR (Variable Bit Rate) traffic between radio mobile terminals and their base stations in a WATM (Wireless ATM) network. After introducing the wireless ATM system model, we describe a new algorithm that enables dynamic slot allocation under TDMA/TDD (Time Division Multiple Access/Time Division Duplex) Media Access Control, making use of UPC (Usage Parameter Control) parameters and traffic characteristics. We show more efficient bandwidth utilization with our proposed algorithm, compared to other conventional algorithms. Moreover, we reveal that our algorithm improves cell transmission delays.

The problem of finding a path with two additive constraints, in particular finding a path that satisfies both the cost and the delay constraints, is called multi-constrained path (MCP) problem in the literature. In this paper, we explore the MCP problem based on the idea of single mixed weight --a mixed weight for each link is first obtained by combining its delay and cost, and then Dijkstra's algorithm is used to find the corresponding shortest path. Given two infeasible paths, it can be theoretically proved that a better path can possibly be found if we choose an appropriate parameter to construct the mixed weight. An approximate algorithm is thus proposed to solve the MCP problem. Theoretical analysis demonstrates that this algorithm can make a correct judgment whether there is a feasible path or not with a very high probability even in the strict case where the delay bound is between the delays of the least delay path and the least cost path, while the cost bound is between the costs of the two paths. On the other hand, the time complexity of this algorithm is very small since it only needs to execute Dijkstra's algorithm a limited number of times. The excellent performance of the proposed algorithm is verified by a large number of experiments on networks of different sizes.

This paper describes the concept of an adaptive network, that is, a network environment that can rapidly and autonomously adapt its behavior according to network conditions and traffic status. The user interface of the adaptive network can access any resource in the network as a memory-mapped I/O device, as if it were attached to the local bus of the user's PC. This network concept has several benefits. From the application development viewpoint, no network related programming is required, and applications do not have to be modified even if the network topologies and protocols are changed. Network maintenance and upgrading can be done anytime without having to worry about the application users, because the network itself is concealed from the applications. In addition, the reconfigurable hardware technology functions as an autonomous network control through the use of a lower-layer protocol. We developed a testbed that makes heterogeneous resources available to users and used it to demonstrate the feasibility of our concept by implementing and running some applications over it.

In this letter, we present an output feedback controller design technique for uncertain discrete time systems with multiple time-delays. Based on Lyapunov second method, a sufficient condition for the robust stability of the system with a dynamic controller is derived in terms of the linear matrix inequality (LMI) with respect to design variables. The solutions of the LMIs can be easily obtained using existing efficient convex optimization techniques.

The advances in services using the present Internet mean that there will be increasing demand for Video on Demand (VoD) on the Internet in the near future. However, because of the best-effort characteristics of the Internet, it is important to suppress the degradation of communication quality caused by packet dropping when Internet traffic is congested. This paper focuses on MPEG transmission over the Internet, and suitable control mechanisms are established for an acceptable Quality of Service (QoS) improvement through detailed evaluation. Packets are classified using a frame-based scheme. The server applies the proposed End-to-End control scheme and shuffles the order of packets to avoid burst dropping, and may omit selected packets belonging to certain frames prior to forwarding. At the intermediate routers, transferred packets are transmitted according to Round Robin (RR) or Weighted Round Robin (WRR) scheduling, and are dropped statistically using selective Random Early Detection (RED) corresponding to frame attributes when there is congestion. We evaluate the proposed performance of transmission method using both computer simulations and empirical measurements of picture quality. The results show that when the traffic volume cannot be estimated in the intermediate routers, the combined use of RR, shuffling and conditional RED is effective, and when the traffic volume can be estimated, the combination of WRR, rate control and RED is effective.

In order for a service discipline to be used for guaranteed service networks at very high speed, its overall implementation must be scalable while it provides as wide a network schedulability region as possible. From this point of view, GPS-based service disciplines provide a narrow network schedulability region while EDF-based disciplines suffer from the implementation complexities of rate-controllers and admission control. Alternatively, although service disciplines based on service-curves can provide a wider network schedulability region than GPS-based and EDF-based disciplines, they may have even worse implementation complexities than EDF-based disciplines. In this paper, we propose to employ a service discipline based on our specific service-curves. We show that our service discipline has comparable implementation complexity to GPS-based disciplines while providing the same wide network schedulability region that EDF-based disciplines can provide. In fact, this service discipline is an SCED service discipline proposed in [14]. However, our specific service-curves provide the SCED service discipline with the same network schedulability region that EDF-based disciplines can provide, O(1) complexity for deadline calculation, and O(N) complexity for admission control where N is the number of sessions.