• The Journal of Korean Institute of Communications and Information Sciences
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• v.37B no.11
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• pp.1004-1013
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• 2012

The legacy voice broadcast systems are used to broadcast the voice over an entire space or a specific zone. these broadcast systems generate unnecessary noise and waste of resources. In this paper, we propose a ubiquitous voice message broadcast system called uPaging, by combining the technique of location-awareness and the voice message delivery service in ubiquitous sensor network environment. In uPaging system, the wire/wireless hybrid network is used to implement the network system. Also, in order to actualize the location-awareness service, we use the Bidirectional Location ID-Exchange protocol was suggested by our previous research. the uPaging system can deliver the voice to a selected user or the location in which the user is present by this location awareness.

• ETRI Journal
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• v.23 no.2
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• pp.52-60
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• 2001

In this paper, a medium access control protocol is proposed for integrated voice and data services in wireless local networks. Uplink channels for the proposed protocol are composed of time slots with multiple spreading codes per slot based on slotted code division multiple access (CDMA) systems. The proposed protocol uses spreading code sensing and reservation schemes. This protocol gives higher access priority to delay-sensitive voice traffic than to data traffic. The voice terminal reserves an available spreading code to transmit multiple voice packets during a talkspurt. On the other hand, the data terminal transmits a packet without making a reservation over one of the available spreading codes that are not used by voice terminals. In this protocol, voice packets do not come into collision with data packets. The numerical results show that this protocol can increase the system capacity for voice service by applying the reservation scheme. The performance for data traffic will decrease in the case of high voice traffic load because of its low access priority. But it shows that the data traffic performance can be increased in proportion to the number of spreading codes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.1
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• pp.92-100
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• 2016

It is informed that the Voice over LTE(VoLTE) which serves voice and message on IP networks is better in terms of economies of scale than the legacy voice service on 2G/3G circuit-switched networks because of its technological and cost efficiency. In addition, services of voice and data are running on a single LTE network and as a result VoLTE has the more economies of scope. But, there is no study about how much technology-efficiency VoLTE has compared to circuit-based voice service and how much voice charge can be reduced as VoLTE grows up. This paper analyzes empirically cost-efficiency of VoLTE against circuit-based voice service and quantifies the reduction of voice charge as 2G/3G voice traffic shifts to VoLTE. The results describe the first is that the average cost of the total voice traffic rises shortly just after the investment of LTE network for providing VoLTE but it will soon have a capacity available to reduce the charge due to VoLTE's outstanding cost efficiency on the assumption that voice traffic is fixed, and the second is that the charge can be cut to 60% of the current rate in case of all the voice traffic moves to VoLTE. The latter proves partially the validation of data-focusing pricing plan. Our results are expected to become basic data for network operators' establishing pricing strategies and for policy makers' inducing price cutting.

• The Journal of the Acoustical Society of Korea
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• v.22 no.3
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• pp.242-249
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• 2003

When special voice control application services (VCS) such as voice-controlled web browsing or voice-controlled stock transactions are introduced in cellular systems, a channel quality better than that for ordinary voice communications service (OVS) is necessary in order to keep a suitable grade of VCS. To avoid ai. congestion, calls are normally admitted if there exists a channel-processing resource not occupied by other calls in the base as well as the interference level at the receiver is not higher than a predefined threshold. The threshold is usually 10㏈ noise-rise over the background noise level for voice communications service. When the base admits VCS attempts in exactly the same manner as it handles OVS calls. the same fraction of those will be not successful in taking the channel and then blocked. If the same noise-rise threshold is used as 10 ㏈, however, the admitted VCS calls might suffer from bad channel qualify and finally be dropped. From the user's point of view, the forced termination of ongoing calls is significantly undesirable than blocking new call attempts. When using a lower noise-rise threshold for VCS. on the other hand, the blocking probability of VCS gets higher than that of OVS. In this paper, a call admission policy that gives a priority to VCS is considered in order to reduce the blocking probability and keep an adequate channel quality.

• Proceedings of the IEEK Conference
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• 2002.06a
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• pp.307-310
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• 2002

This paper is proposed due to the fact that resolving domain name with conventional input method to PDA is not so convenient. The substance of this paper is that a user Pronounces tile character which represents tile domain name and that vocalized character is transferred through the GATEWAY, where DNS service can be received in return. PDA receives, compress and send the voice to the GATEWAY, Then, the GATEWAY uncompress, recognizes the voice, converts to characters, search for the mapping entry After mapping to the mapping entry, the GATEWAY sends the DNS request. Combining two entities makes DNS service based on the human voice possible.

• Journal of the Ergonomics Society of Korea
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• v.21 no.1
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• pp.51-65
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• 2002

As the method of providing information is getting multimedia, the synthetic voice is used in not only CTI(Computer Telephony Integration), information service for the blind, but also applications on internet. But properties of synthetic voice, such as speech rate, pitch, timbre and so on, are not adjusted to customers' preference but providers' preference. In order to consider customers' preference, this study proposed four subjective factors of voice through the evaluation of voice using the method of sensibility ergonomics. And the relation synthetic voice to be agreeable to the ear with emotional images was formulated as a fuzzy model. Consequently, this study proposed the speech rate and pitch of synthetic voice which is agreeable to the ear.

• Journal of Information Technology Applications and Management
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• v.14 no.4
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• pp.221-233
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• 2007

Which factors should influence consumer consideration to subscribe to Voice over Internet Protocol (VoIP)? Policy issues, managerial concerns, and demographic variables are possible factors. This paper discusses policy and managerial issues regarding VoIP adoption. A model that explains VoIP adoption is proposed and tested. This study analyzes a survey of 750 prospective VoIP users in Korea. The testing is accompanied by logistic regression and discriminant analysis. The results show that trust in VoIP, relative comparison of Quality to fixed service, numbering plan, satisfactions of call Quality and customer services on both fixed and mobile services have impacts on the adoption of VoIP. Implications for VoIP providers and policy makers are presented.

• ETRI Journal
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• v.28 no.3
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• pp.311-319
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• 2006

This paper presents a terminal-assisted frame-based packet reservation multiple access (TAF-PRMA) protocol, which optimizes random access control between heterogeneous traffic aiming at more efficient voice/data integrated services in dynamic reservation TDMA-based broadband access networks. In order to achieve a differentiated quality-of-service (QoS) guarantee for individual service plus maximal system resource utilization, TAF-PRMA independently controls the random access parameters such as the lengths of the access regions dedicated to respective service traffic and the corresponding permission probabilities, on a frame-by-frame basis. In addition, we have adopted a terminal-assisted random access mechanism where the voice terminal readjusts a global permission probability from the central controller in order to handle the 'fair access' issue resulting from distributed queuing problems inherent in the access network. Our extensive simulation results indicate that TAF-PRMA achieves significant improvements in terms of voice capacity, delay, and fairness over most of the existing medium access control (MAC) schemes for integrated services.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1988.10a
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• pp.12-16
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• 1988

본 논문에서는 안내원의 작업처리시간을 줄이기 위한 방법으로 DDX-1A를 이용한 자동호 분배 장치에 자도음성안내 기능을 구현하기 위한 T-level Prncessor 인 PCP (Protocol Convert Processor) VCP(Voice Contro Processor)와 B-level Processor Avru(voice Response)와 B-level Processor AVRU(Automatic Voice Response Unit)의 H/W 기능 및 상호 interface 에 관하여 고찰한다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4
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• pp.42-47
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• 2001

In this paper, we propose an interpreter, which recognizes the VoiceXML markups, verifies the validation of the document, and interprets the VoiceXML documents using DI parser and the generated AST by the parser. The VoiceXML interpreter consists of DI parser and executor, and the DI parser uses recursive descent parsing technology, and the executor uses FIA (Form Interpretation Algorithm) proposed by VXML forum. This system uses the Java language in order to develop the runtime environment for VoiceXML efficiently, thus this system has portability.