• 대한임베디드공학회논문지
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• 제2권1호
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• pp.43-52
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• 2007

IEEE 802.15.4 specification which defines low-rate wireless personal area network(LR-WPAN) has application to home or building automation, remote control and sensing, intelligent management, environmental monitoring, and so on. Recently, it has been considered as an alternative technology to provide multimedia services such as automation via voice recognition, wireless headset and wireless camera for surveillance. In order to evaluate capability of voice traffic on the IEEE 802.15.4 LR-WPAN, we supposed two scenarios, voice traffic only and coexistence of voice and sensing traffic. For both cases we examined delay and packet loss rate in case of with and without acknowledgement, and various beacon period varying with beacon and superframe order values. In LR-WPAN with voice devices only, total 5 voice devices could be applicable and in the other case, i.e., coexisted cases of voice and sensor devices, a voice device was able to coexist with about 60 sensor devices.

• 정보처리학회논문지C
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• 제8C권5호
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• pp.657-666
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• 2001

본 논문에서는 글로벌 캐리어들이 고객 중심으로 음성 트래픽을 효율적으로 처리 및 관리할 수 있도록 CDR을 이용하여 음성 트래픽 관리 항목들을 제안하였으며, 관리 항목을 산출하기 위한 계산 수식들을 정의하였다. 이러한 관리 항목들을 웹 인터페이스 모듈, 분석 모듈, 데이터 수집 모듈, 데이터베이스 관리 모듈로 분리하여 시스템을 설계하였으며 웹 기술을 이용하여 시스템을 구현함으로써 가용성 및 사용의 편의성을 높였다. 또한, 규정된 관리 항목의 유효성을 검증하기 위하여 실제로 글로벌 캐리어에 의해 수집된 CDR을 바탕으로 관리 항목들을 실험하였다. 제안된 웹 기반 음성 트래픽 관리 시스템은 글로벌 캐리어들에게 네트워크 정보 수집, 장애 판단 및 원인 해결, 고객의 성향 분석을 통한 보다 질 높은 서비스 제공 등 도움을 줄 것으로 기대된다.

• 한국통신학회논문지
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• 제19권2호
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• pp.354-363
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• 1994

본 논문은 광대역 ISDN에서 음성 서비스를 효율적으로 제공하는 방법에 관한 연구이다. 음성은 그 특성상 유음구간과 북음구간으로 나누어지며, 음성활성검출에 의하여 실제로 말을 하는 동안만 전송이 이루어질 수 있도록 음성 트래픽을 발생한다. 본 논문에서는 ATM통신망에서 음성을 음성활성검출과 삽입(Embedded) ADPCM으로 코딩하고, 멀티플렉서에서 셀 폐기를 통하여 트래픽을 제어하는 알고리즘에 관하여 연구하였다. 트래픽 제어는 버퍼에 임계값을 주어, 버퍼의 상태가 임계값을 초과하는 경우에 낮은 우선순위를 갖는 음성 셀을 폐기하는 셀 폐기 알고리즘을 사용하였다. 셀 손실 확률, 큐 크기, 평균지연등을 성능분석 파라메터로 설정하고, 트래픽 제어 알고리즘의 성능을 평가하기 위하여 컴퓨터 시뮬레이션하였다. 이를 통하여 센 폐기방식에 의한 트래픽 제어가 음성의 질을 많이 저하시키지 않으면서도 트래픽 제어를 하지 않을 때에 비하여 전송 대역 이득을 향상시킨다는 것을 확인하였다.

• 한국경영과학회:학술대회논문집
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• 한국경영과학회/대한산업공학회 1999년도 춘계공동학술대회:정보화시대의 지식경영
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• pp.371-372
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• 1999

• ETRI Journal
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• 제20권2호
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• pp.171-191
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• 1998

In this paper, we propose a new heuristic design algorithm for the virtual path (VP)-based ATM network with multiple traffic classes, in which QoS constraints associated with traffic class are taken into account. The minimum bandwidth of VP required to carry given amount of traffic is obtained by utilizing an equivalent bandwidth concept, and the route of each VP is placed so that the network cost is minimized while the QoS requirement is fulfilled To evaluate our design algorithm, we consider two kinds of traffic: voice traffic as low speed service and still picture traffic as high speed service. Through numerical examples, we demonstrate that our design method can achieve an efficient use of network resources, which results in providing a cost-effective VP-based ATM network.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(3)
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• pp.145-148
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• 2000

ATM network is considered as best suitable technology for multimedia service in various aspects such as bandwidth, capability of traffic expandability and so on. In this paper, we suggested a buffer management algorithm in ATM network to improve overall network performance with threshold and auxiliary buffer whose input consists of superpositiI on of voice and multimedia data traffic. To evaluate the proposed buffer management algorithm simulations are executed with four priorities, that is delay and loss priorities and the results are proved that network throughput is improved better than the existing partial buffer method.

• 한국경영과학회지
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• 제24권1호
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• pp.49-58
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• 1999

A new channel access function is proposed to enhance the performance of the Joint CDMA/PRMA. It is obtained in consideration of the number of terminals in reservation mode and the number of terminals in contention mode whose probability distribution is estimated by applying pseudo Bayesian approach. Simulation results show that the performance of the Joint CDMA/PRMA can be improved by applying new channel access function under voice-only traffic and mixed voice/random-data traffic.

• 경영과학
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• 제10권1호
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• pp.107-125
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• 1993

Modern telecommunication switching systems are SPC(Stored Program Control) machines handling voice, data and other kinds of traffic, in an environment which tends to be fully digital switching and transmission. The throughput of such systems is determined by the real time capacity of its centralized or distributed control processors and by the traffic capacity of the switching network. Designers must verify the traffic and call processing capacity of the switching system and check its performance under traffic load before it is put into service. Verification of traffic and call processing capacity of switching systems is one of the problems treated by teletraffic studies ; teletraffic studies are based on stochastic process, queueing theory, simulations and other quantitative methods of decision making. This study suggests the general methodology to evaluate the throughput and performance of the ISDN switching system. TDX-10 ISDN switching system are employed to give illustrative examples of the methodologies discussed in this study.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제14권2호
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• pp.473-494
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• 2020

To provide a guaranteed Quality of Service (QoS) to real-time traffic in High-Speed Downlink Packet Access (HSDPA) core network, we proposed an enhanced mechanism. For an enhanced QoS, a Class-Based Low Latency Fair Queueing (CBLLFQ) packet scheduling algorithm is introduced in this work. Packet classification, metering, queuing, and scheduling using differentiated services (DiffServ) environment was the points in focus. To classify different types of real-time voice and multimedia traffic, the QoS provisioning mechanisms use different DiffServ code points (DSCP).The proposed algorithm is based on traffic classes which efficiently require the guarantee of services and specified level of fairness. In CBLLFQ, a mapping criterion and an efficient queuing mechanism for voice, video and other traffic in separate queues are used. It is proved, that the algorithm enhances the throughput and fairness along with a reduction in the delay and packet loss factors for smooth and worst traffic conditions. The results calculated through simulation show that the proposed calculations meet the QoS prerequisites efficiently.

• Journal of Communications and Networks
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• 제10권3호
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• pp.287-296
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• 2008

The IEEE 802.11 wireless standard uses the carrier sense multiple access with collision avoidance (CSMA/CA) as its MAC protocol (during the distributed coordination function period). This protocol is an adaptation of the CSMA/CD of the wired networks. CSMA/CA mechanism cannot guarantee quality of service (QoS) required by the application because orits random access method. In this study, we propose a new MAC protocol that considers different types of traffic (e.g., voice and data) and for each traffic type different priority levels are assigned. To improve the QoS of IEEE 802.11 MAC protocols over a multi-channel CSMA/CA, we have developed a new admission policy for both voice and data traffics. This protocol can be performed in direct sequence spread spectrum (DSSS) or frequency hopping spread spectrum (FHSS). For voice traffic we reserve a channel, while for data traffic the access is random using a CSMA/CA mechanism, and in this case a selective reject and push-out mechanism is added to meet the quality of service required by data traffic. To study the performance of the proposed protocol and to show the benefits of our design, a mathematical model is built based on Markov chains. The system could be represented by a Markov chain which is difficult to solve as the state-space is too large. This is due to the resource management and user mobility. Thus, we propose to build an aggregated Markov chain with a smaller state-space that allows performance measures to be computed easily. We have used stochastic comparisons of Markov chains to prove that the proposed access protocol (with selective reject and push-out mechanisms) gives less loss rates of high priority connections (data and voices) than the traditional one (without admission policy and selective reject and push-out mechanisms). We give numerical results to confirm mathematical proofs.