• Animal cells and systems
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• v.6 no.3
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• pp.221-225
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• 2002

This study was conducted to find out how young black-tailed gulls (Larus crassirostris) recognize adult voice signals after hatching. For the experiment, adult voice recorded in the natural environment was played back at controlled intervals and intensity (dB) to 15 young gulls that were artificially hatched in the laboratory. The chirirah call frequency of young gulls increased as the intensity of the mew call increased. The chirirah response of the control group was highest to the mew call at intervals of 1.8s. The adult long ca11 and alarm call also showed similar results to the mew call when the interval and intensity were manipulated similar to the mew call. Based on the results of this experiment, it is assumed that the young black-tailed gulls recognize adult voice signals based on the simple structure of adult voice signals, that is, the interval and intensity of the voice.

• Proceedings of the IEEK Conference
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• 2000.11c
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• pp.161-164
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• 2000

This paper presents an implementation of internet gateway system, named AX-2000 and call simulator. AX-2000 plays a important role in internet telephony technology and is composed of various board such as MPU, AVU. DVU, AMU. FXS, FXO, EM. Also AX-2000 supports G.729.a, G.723.1 for voice compression, G3 FAX Relay(T.38) and H.323. A capability of AX-2000 is 8 analog voice channel or 30 digital voice channel. For functional verification of AX-2000 voice interface, call simulator is designed. The call simulator makes actual call path between SUT(system under test) and reference AX-2000 system, then through call path examines functions of voice interface.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.27-36
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• 2002

VoiceXML is a standard dialog mark-up language for the neat generation voice applications. The current VoiceXML 1.0 specification is silent on who place outbound calls for push-based voice applications. The push-barred voice applications become very important in modern information systems such as CRM. In this paper, we design and implement an extended VoiceXML platform that supports both inbound and outbound voice information services. We also extend the VoiceXML DTD so as to be able to inbound/outbound fax based on Call Control Requirements of W3C.

• MALSORI
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• no.53
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• pp.171-183
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• 2005

Call Control eXtensible Markup (CCXML) is a standard language that supports a call control of voice dialog systems such as VoiceXML based systems. CCXML allows developers to handle telephony calls in an easy way without deep knowledge about telephony networks and their switching systems.We design and implement a call control markup interpreter. At the implementation, we use a Dialogic JCT-LS board, but, by designing a wrapping class for CTI (computer telephony board) features, the interpreter can easily adopt other CTI boards. We also design and implement event-based interaction scheme between the interpreter and voice dialog systems. For verifying the interaction scheme, we implement a simple voice dialog system.

• MALSORI
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• no.49
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• pp.77-94
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• 2004

In this paper, new voice message system supporting massive outbound call is proposed. Basic idea of the proposed system is to pre-process all the text-to-speech conversion process, mixing of text and attached music file and to store the results of pre-process in the cache server which is connected to the IVR. New voice message system is optimized for the voice message system supporting massive outbound call by distributing the load of the web server caused by server-side script implementation which is accessing database and generating dynamic Voice XML document over client module and server module of web server. The proposed voice message system was test-deployed in one domestic voice message application service provider and it is shown that proposed voice message system reduced the response latency problem of test-bed voice message system.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.42-50
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• 2003

In this paper we propose a method of evaluating the performance of a Softswitch that provides call control to voice services in NGN (next generation network). First, we describe the architecture for voice services in NGN and anatomize the call control processes such as call initiation, call re-initiation and call release of a voice connection. kiter that we propose a method of estimating appropriate server capacity of the Softswitch using approximate queuing model. Via numerical experiments we illustrate the implication of the work

• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.26-30
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• 2006

Internet phone service is a new service technology that provides voice call services through Internet not through the pre-existing PSTN. It enables a cheap voice call service regardless of distance. We may expect that the Internet phone service may substitute for the voice call service through the PSTN, but not in a short period. There are several problems to be solved for this transition, such as, voice call quality, numbering scheme, billing, standardization, and support of several functions. In this paper, we provided and designed a UA (User Agent) that can support functions regarding voice call, such as call forwarding, auto-connection, call rejection and restriction of individual call, using SIP (Session Initiation Protocol) which is proposed by SIP-Working Group as the standard Internet phone service management protocol.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.111-116
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• 2016

According to the Ministry of Science, ICT and Future Planning's survey of information gap, smartphone retention rate of disabled people stayed in one-third of non-disabled people, the situation is significantly less access to information for people with disabilities than non-disabled people. In this paper, we develop an application, CallHelper, that helps to be more convenient to use mobile voice calls to the auditory disabled people. CallHelper runs automatically when a call comes in, translates caller's voice to text output on the mobile screen, and displays the emotion reasoning from the caller's voice to visualize emoticons. It also saves voice, translated text, and emotion data that can be played back.

• Journal of Information Processing Systems
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• v.2 no.2
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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.

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.3
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• pp.208-212
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• 2008

This paper proposes a model of a voice communication using bluetooth and a mobile, and implements a hands-free for this model. A Proposed model uses bluetooth for a voice network, and is possible to share a voice with a great number of people using preemptive algorithm proposed in this paper. There are two types of models. One is a model only using a hands-free, the other is a model using a mobile. Second model uses a scatternet and a call forwarding service for a call on a voice communication. In case of using a scatternet, scatternet is composed of a piconet of a voice communication and a piconet of a call. In case of using a call forwarding service, the mobiles share information of each others before formation of a network.