• Asia pacific journal of information systems
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• 제28권4호
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• pp.320-349
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• 2018

Purpose: The application of IoT is finding continuous acceptance in our daily lives, particularly, smart speakers are making life easier and convenient for consumers. This research aims to develop and test an integrated model of factors influencing consumer's adoption of voice-enabled IoT devices. Design/methodology/approach: Based on the VAM, an integrated voice-enabled IoT device adoption model is proposed. Gender differences on five constructs relating with perceived value (perceived usefulness, perceived enjoyment, perceived security risk, perceived technicality and perceived cost) was also examined through PLS-MGA technique. The usage experience of consumers was also controlled in the integrated VAM. Findings: Result shows that Perceived-Usefulness, Perceived-Enjoyment and Perceived-Cost have a strong effect on Perceived-Value. However, Perceived-Technicality and Perceived-Security-Risk are non-influential and have no significant effect on PV. Additionally, Perceived-Value and Social-Influence plays a significant role in predicting adoption intention. Gender differences also exist in consumers perception of usefulness, enjoyment and cost. In comparison to the basic value-based adoption model, the integrated model provides more insight on consumers adoption of voice-enabled IoT devices. Originality/value: Using an integrated model, this study is one of the first scholarly attempt at modelling the influential factors for adopting smart speakers i.e., voice-enabled IoT devices, with implications for improved adoption.

• Journal of Information Processing Systems
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• 제2권2호
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• pp.107-113
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• 2006

It is essential to guarantee a handoff dropping probability below a predetermined threshold for wireless mobile networks. Previous studies have proposed admission control policies for integrated voice/data traffic in wireless mobile networks. However, since QoS has been considered only in terms of CDP (Call Dropping Probability), the result has been a serious CBP (Call Blocking Probability) unfairness problem between voice and data traffic. In this paper, we suggest a new admission control policy that treats integrated voice and data traffic fairly while maintaining the CDP constraint. For underprivileged data traffic, which requires more bandwidth units than voice traffic, the packet is placed in a queue when there are no available resources in the base station, instead of being immediately rejected. Furthermore, we have adapted the biased coin method concept to adjust unfairness in terms of CBP. We analyzed the system model of a cell using both a two-dimensional continuous-time Markov chain and the Gauss-Seidel method. Numerical results demonstrate that our CAC (Call Admission Control) scheme successfully achieves CBP fairness for voice and data traffic.

• ETRI Journal
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• 제23권2호
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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.

• Journal of information and communication convergence engineering
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• 제5권2호
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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.

• 한국통신학회논문지
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• 제19권9호
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• pp.1748-1758
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• 1994

최근에 이동통신의 수요가 급격히 증가하고 있으며, 음성뿐만 아니라 종합정보통신망(ISDN)에서 제공되는 데이터, 팩스, 영상 등의 비음성계 서비스가 요구되고 있다. 따라서 본 논문에서는 PCS 시스템의 기본 핵심기술로 활용될 수 있는 디지털 셀룰러 CDMA 시스템용 음성/데이터 집적 서비스 방안들을 제시하고 컴퓨터 심뮬레이션을 통하여 각 집적 서비스 방안의 성능을 비교 분석한다. 컴퓨터 시뮬레이션 결과, CDMA의 한 논리채널을 여러 단말기가 할당받아 PRMA 방식으로 경쟁하는 PR-CDMA 음성/데이타 집적 서비스 방안이 CDMA 방식의 단점인 고정 채널갯수의 한계와 채널이용율의 저하를 극복할 수 있어 전용단말기나 음성/데이타 집적단말기를 이용한 광대역 CDMA 음성/데이타 집적서비스 방안 보다 모든 면에서 우수한 성능을 나타냄을 알 수 있었다.

• 음성과학
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• 제12권4호
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• pp.77-92
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• 2005

Voice is made by systemic interaction of respiration, vocalization, articulation and resonation. There is no existing multi-channel voice analysis system to assess voice and respiration simultaneously. The most existing systems consist of. vocal fold vibration measurement such as stroboscopy, EGG (electroglottography) or laryngeal electromyography and voice analysis system. Since respiration has close relationship with voice simultaneous analysis of both vocal cord vibration and respiration are essential. In this study, a four channel integrated system are developed for acoustic analysis through microphone, vocal fold vibratory analysis using EGG, and respiratory analysis using two channel RIP (respiratory inductive plethysmography).

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2004년도 ICCAS
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• pp.1342-1345
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• 2004

An analytical of throughput and delay on reverse link in a cellular slotted CDMA system with integrated voice and World Wide Web (WWW) traffic is presented in this paper. Our results show that the Gaussian approximation it was found that the voice traffic can have a significant impact on the WWW traffic. It was also found that the different mobiles velocity and number of resolvable paths is impact each other the system performance.

• Journal of information and communication convergence engineering
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• 제8권5호
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• pp.513-518
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• 2010

This paper proposes a medium access control protocol for integrated voice and data services in slotted CDMA systems. The proposed protocol, which is named as RCSSMA (Reservation Code and Status Sensing Multiple Access), adopts a code reservation and status sensing schemes. RCSSMA protocol gives higher access priority to the voice traffic than data traffic for reducing the packet dropping probability. The voice terminal reserves an available spreading code to transmit voice packets during a talkspurt, whereas the data terminal transmits a packet over one of the available spreading codes that are not reserved by the voice terminals. In this protocol, the voice packets never contend with the data packets. Packet dropping probability and average data packet transmission delay are analyzed using a Markov chain model.

• 한국통신학회논문지
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• 제17권3호
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• pp.277-287
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• 1992

본 논문에서는 음성 트래픽을 동기 모드로, 데이타 트래픽을 비동기 모드로 집적시켜 전송하는 FDDI 통신망에 대하여 수학적 분석과 시뮬레이션을 수행하여 성능을 분석 한다. 음성과 데이타의 평균 패킷 대기 시간을 구하기 위하여, 음성은 Marcov 모델에서 발음 부분과 묵음 부분이 교대로 나타나는 모델로 하며, 데이타는 TRT와 TTRT의 차이 시간동안 전송하는 모델로 하는 경우, 음성과 데이타의 패킷 대기 시간을 구하여 동기모드와 비동기 모드 사이의 전송관계에 대하여 연구하였다. 연구 결과 동기/ 비동기 집적 방식으로 음성과 데이타를 집적하여 전송하여도 음성 채널 용량의 제약없이 데이타를 집적 시킬 수 있었으며, 데이타 전송도 작은 대기 시간내에 신속하게 전송할 수가 있었다.

• 한국정보과학회논문지:정보통신
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• 제29권2호
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• pp.141-155
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• 2002

본 논문에서는 기지국과 단말기간의 전파지연이 비교적 짧은 마이크로셀 환경의 패킷 CDMA망에서 음성과 데이타를 통합 서비스하기 위한 MAC 프로토콜을 제안한다. 제안한 프로토콜을 위한 시스템의 상향링크 채널은 프레임을 구성하는 타임 슬롯과 각 슬롯별 확산코드에 의하여 제공된다. 지연에 민감한 음성 트래픽에 더 높은 우선순위를 부여하여, 음성 트래픽인 경우에는 발음구간 동안 임의의 확산코드를 예약하여 서비스된다. 반면, 데이타 트래픽은 매 패킷이 발생할 때마다 기지국으로부터 수신한 현재 슬롯의 예약상태 정보를 기반으로 서비스된다. 성능분석의 결과, 본 논문에서 제안한 기법은 음성과 데이타 패킷간의 충돌이 전혀 발생하지 않으므로 주어진 패킷 누락률 범위 내에서 수용 가능한 최대 음성 단말기 수를 증가시킬 수 있었다.