• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.697-710
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• 2004

Differentiated services (DiffServ) model has been prevailed as a scalable approach to provide quality of service in the Internet. However, there are difficulties in providing the guaranteed service in terms of end-to-end systems since differentiated services network considers quality of service of aggregated traffic due to the scalability and many researches have been mainly focused on per hop behavior or a single domain behavior. Furthermore quality of service may be time varying according to the network conditions. In this paper, we study dynamic class mapping mechanism to guarantee the end-to-end quality of service for multimedia traffics with the minimum network cost over differentiated services network. The proposed algorithm consists of an effective implementation of relative differentiated service model, quality of service advertising mechanism and dynamic class mapping mechanism. Finally, the experimental results are provided to show the performance of the proposed algorithm.

• Journal of Korea Multimedia Society
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• v.6 no.7
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• pp.1219-1230
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• 2003

We analyze an integrated voice/data CDMA system, where the whole channels are divided into voice prioritized channels and voice non-prioritized channels. For real-time voice service, a preemptivc priority is granted in the voice prioritized channels. And, for delay-tolerant data service, the employment of buffer is considered. On the other hand, the transmission permission probability in best-effort packet-data service is controlled by estimating the residual capacity available for users. We build a 2-dimensional markov chain about prioritized-voice and stream-data services and accomplish numerical analysis in combination with packet-data traffic based on residual capacity equation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7C
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• pp.691-699
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• 2006

Despite of the in-depth effort to cantrol the variability in IP networks, quality of service (QoS) is still not guaranteed in the IP networks. Thus, it is necessary to deal with the audible artifacts caused by packet lasses. To overcame the packet loss problem, most speech coding standard have their own embedded packet loss concealment (PLC) algorithms which adapt extrapolation methods utilizing the dependency on adjacent frames. Since many low bit rate CELP coders use predictive schemes for increasing coding efficiency, however, error propagation occurs even if single packet is lost. In this paper, we propose an efficient PLC algorithm with consideration about the speech characteristics of lost frames. To design an efficient PLC algorithm, we perform several experiments on investigating the error propagation effect of lost frames of a predictive coder. And then, we summarize the impact of packet loss to the speech characteristics and analyze the importance of the encoded parameters depending on each speech classes. From the result of the experiments, we propose a new PLC algorithm that mainly focuses on reducing the error propagation time. Experimental results show that the performance is much higher than conventional extrapolation methods over various frame erasure rate (FER) conditions. Especially the difference is remarkable in high FER condition.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.12B
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• pp.2320-2334
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• 1999

In this paper, we propose multiple IS-MMBS service schemes for very low power and micro-size RMIMS (radio-interfaced micro information monitoring system) terminals. In MMBS service area, when new arrival RMIMS terminals have real-time traffic characteristic or large traffic bandwidth, only single IS-MMBS service scheme can not guarantee RMIMS terminal's QoS(quality of service) such as buffer overflow or packet loss. In this case, the proposed multiple IS-MMBS service schemes can be effectively used for QoS service of RMIMS terminal. According to clustering method of RMIMS terminals and MMBS segment method, the proposed schemes can be divided into terminal segment method, region segment method, application based segment method, traffic type based segment method, overlapping segment method and hybrid segment method

• ETRI Journal
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• v.30 no.2
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• pp.227-237
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• 2008

In this paper, an algorithm that provides absolute and proportional differentiation of packet delays is proposed with the objective of enhancing quality of service in future packet networks. It features an adaptive scheme that adjusts the target delay for every time slot to compensate the deviation from the target delay, which is caused by prediction error on the traffic to arrive at the next time slot. It predicts the traffic to arrive at the beginning of a time slot and measures the actual arrived traffic at the end of the time slot. The difference between them is utilized by the delay control operation for the next time slot to offset it. Because the proposed algorithm compensates the prediction error continuously, it shows superior adaptability to bursty traffic and exponential traffic. Through simulations we demonstrate that the algorithm meets the quantitative delay bounds and is robust to traffic fluctuation in comparison with the conventional non-adaptive mechanism. The algorithm is implemented with VHDL on a Xilinx Spartan XC3S1500 FPGA, and the performance is verified under the test board based on the XPC860P CPU.

• Korean Journal of Optics and Photonics
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• v.27 no.6
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• pp.212-217
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• 2016

In connection-oriented data-transport services, data loss can occur when the service experiences a problem on its end-to-end path. To promptly resolve this problem, the data-switching systems providing the service should quickly modify their internal configurations distributed among different places in each system. This is performed through a sequence of problem (event) recognition, sharing, and handling procedures among multiple control processors in the system. This paper proposes a method for event sharing and messaging between control processors, to improve the time determinacy of event processing. This method simplifies runtime event sharing and minimizes the time variability caused by the event data, resulting in a decrease in the latency time in processing global events. The proposed method lessens the latency time of global event processing by about 40%, compared to general methods, for 738 internal path changes.

• ETRI Journal
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• v.29 no.2
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• pp.201-211
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• 2007

This paper focuses on the scheduling problem with the objective of maximizing system throughput, while guaranteeing long-term quality of service (QoS) constraints for non-realtime data users and short-term QoS constraints for realtime multimedia users in multiclass service high-speed uplink packet access (HSUPA) systems. After studying the feasible rate region for multiclass service HSUPA systems, we formulate this scheduling problem and propose a multi-constraints HSUPA opportunistic scheduling (MHOS) algorithm to solve this problem. The MHOS algorithm selects the optimal subset of users for transmission at each time slot to maximize system throughput, while guaranteeing the different constraints. The selection is made according to channel condition, feasible rate region, and user weights, which are adjusted by stochastic approximation algorithms to guarantee the different QoS constraints at different time scales. Simulation results show that the proposed MHOS algorithm guarantees QoS constraints, and achieves high system throughput.

• Journal of information and communication convergence engineering
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• v.5 no.2
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• pp.88-92
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• 2007

In this paper, an SS-RRA protocol that is based on Code Division Multiple Access is proposed and analyzed under the integrated voice and data traffic load. The backward logical channels consist of slotted time division frames with multiple spreading codes per slot. The protocol uses a reservation mechanism for the voice traffic, and a random access scheme for the data traffic. A discrete-time, discrete-state Markov chain is used to evaluate the performance. The numerical results show that the performance can be significantly improved by a few distinct spreading codes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.5A
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• pp.447-451
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• 2010

We propose an advanced BMPA (balanced multimedia packet allocation) scheme. The proposed scheme considers the optimal resource allocation problem in OFDMA system. When NRT (non-real time), RT (real time), multicast packets arrive at the same time in OFDMA system, the advanced BMPA chooses the best sub-carrier as a multicast channel for multicast packets, gives more weight to RT packets and multicast packets than NRT packets and allocates sub-carriers according to the total weight sum of packets in each user. With simulation results, this paper shows that the advanced BMPA scheme ensures QoS (long-term system throughput and multimedia packet delay) in heterogeneous traffic compared with multicast MU-WF (multi-user water-filling) and BMPA schemes.