• IEMEK Journal of Embedded Systems and Applications
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• v.2 no.1
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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.

• The KIPS Transactions:PartC
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• v.8C no.5
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• pp.657-666
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• 2001

In this paper, it is proposed the management items for voice traffic using CDRs so that global carriers can treat and manage the voice traffic for a customer, and defined computational expressions to produce the management items. From them, we have designed the management system, which is composed of web interface module, analysis module, data collection module and database management module, and have improved the availability and convenience of the system using web technologies. In addition, we have tested these items using CDRs in real environments that are collected by the global carrier in order to verify their validity. It is expected that the proposed web based voice traffic management system provide a global carrier with network information collection, fault detection/trouble-shooting and high quality of service through analyzing the characteristics of subscribers.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.2
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• pp.354-363
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• 1994

This paper describes the method that voice traffic is served efficiently in BISDN. Voice is divided into talkspurt and silent period, and it is possible to transmit olny talksurt by the speech activity detection. This paper described the voice traffic control algorithm in the ATM network where cell discarding method is applied to the embedded ADPCM voice data. For traffic control, the cell discarding was used over low priority cells when it overflows the queue threshold. To estimate the efficiency of traffic control algorithm, the computer simuation was performed with cell loss probability, queue length and mean delay as performance parameters. The embedded ADPCM voice coding and cell disscarding resulted in improving the voice cell traffic efficiency and the dynamic control over network congestion.

• Proceedings of the Korean Operations and Management Science Society Conference
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• 1999.04a
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• pp.371-372
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• 1999

• ETRI Journal
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• v.20 no.2
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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.

• Proceedings of the IEEK Conference
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• 2000.06c
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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.

• Journal of the Korean Operations Research and Management Science Society
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• v.24 no.1
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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.

• Korean Management Science Review
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• v.10 no.1
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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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• v.14 no.2
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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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• v.10 no.3
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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.