• Proceedings of the KIEE Conference
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• 2005.07d
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• pp.2771-2773
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• 2005

In network human-robot interaction, human can access services of a robot system through the network The communication is done by interacting with the distributed sensors via voice, gestures or by using user network access device such as computer, PDA. The service organization and exploration is very important for this distributed system. In this paper we propose a new agent-based framework to integrate partners of this distributed system together and help users to explore the service effectively without complicated configuration. Our system consists of several robots. users and distributed sensors. These partners are connected in a decentralized but centralized control system using agent-based technology. Several experiments are conducted successfully using our framework The experiments show that this framework is good in term of increasing the availability of the system, reducing the time users and robots needs to connect to the network at the same time. The framework also provides some coordination methods for the human robot interaction system.

• 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.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.113-118
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• 2005

Softswitch is the core BcN equipment which voice and multimedia switching based on the IP Technologies. It is designed to replace the Class 5(local Exchange) and Class 4(Toll Exchange) switch based on the circuit wired and wireless switching network technologies. Softswitch gets its name because typically it is a software based solution implemented on general purpose computers/servers. While the traditional PSTN switches are rely on dedicated facilities for T and S inter-connection and are designed primarily for voice communications. Packet based Softswitch is divided the control of call and bearer, very different from Public telephone network. Sometimes Call Agent or Media Gateway Controller, a key component in the VoIP solution, is also called Softswitch. This paper will suggest the software architecture of softswitch for performance in call processing part, also suggest the session management model to cover call setup latency.

• Journal of the Korea Society of Computer and Information
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• v.29 no.4
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• pp.23-30
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• 2024

In this paper, we propose a deep learning-based personalized senior care service application. The proposed application uses Speech to Text technology to convert the user's speech into text and uses it as input to Autogen, an interactive multi-agent large-scale language model developed by Microsoft, for user convenience. Autogen uses data from previous conversations between the senior and ChatBot to understand the other user's intent and respond to the response, and then uses a back-end agent to create a wish list, a shared calendar, and a greeting message with the other user's voice through a deep learning model for voice cloning. Additionally, the application can perform home IoT services with SKT's AI speaker (NUGU). The proposed application is expected to contribute to future AI-based senior care technology.

• Journal of Intelligence and Information Systems
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• v.29 no.3
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• pp.267-286
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• 2023

The conversational agents such as AI speakers utilize voice conversation for human-computer interaction. Voice recognition errors often occur in conversational situations. Recognition errors in user utterance records can be categorized into two types. The first type is misrecognition errors, where the agent fails to recognize the user's speech entirely. The second type is misinterpretation errors, where the user's speech is recognized and services are provided, but the interpretation differs from the user's intention. Among these, misinterpretation errors require separate error detection as they are recorded as successful service interactions. In this study, various text separation methods were applied to detect misinterpretation. For each of these text separation methods, the similarity of consecutive speech pairs using word embedding and document embedding techniques, which convert words and documents into vectors. This approach goes beyond simple word-based similarity calculation to explore a new method for detecting misinterpretation errors. The research method involved utilizing real user utterance records to train and develop a detection model by applying patterns of misinterpretation error causes. The results revealed that the most significant analysis result was obtained through initial consonant extraction for detecting misinterpretation errors caused by the use of unregistered neologisms. Through comparison with other separation methods, different error types could be observed. This study has two main implications. First, for misinterpretation errors that are difficult to detect due to lack of recognition, the study proposed diverse text separation methods and found a novel method that improved performance remarkably. Second, if this is applied to conversational agents or voice recognition services requiring neologism detection, patterns of errors occurring from the voice recognition stage can be specified. The study proposed and verified that even if not categorized as errors, services can be provided according to user-desired results.

• Phonetics and Speech Sciences
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• v.4 no.2
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• pp.127-134
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• 2012

The purpose of this study is to present evidence for a particular processing approach based on the language-specific characteristics of Korean. To compare individuals' sentence-comprehension abilities, this study measured the accuracy and reaction times (RT) of 12 aphasic patients (AP) and 12 normal controls (NC) during a sentence-picture matching task. Four versions of a sentence were constructed with the two types of voice (active/passive) and two types of word order (agent-first/patient-first). To examine the effects of increased processing demand, picture stimuli were manipulated in such a way that they appeared immediately after the sentence was presented. As expected, the AP group showed higher error rates and longer RT for all conditions than the NC group. Furthermore, Korean speakers with aphasia performed above a chance level in sentence comprehension, even with passive sentences. Aphasics understood sentences more quickly and accurately when they were given in the active voice and with agent-first order. The patterns of the NC group were similar. These results confirm that Korean adults with aphasia do not completely lose their knowledge of sentence comprehension. When the processing demand was increased by delaying the picture stimulus onset, the effect of increased processing demands on RT was more pronounced in the AP than in the NC group. These findings fit well with the idea that the computational system for interpreting sentences is intact in aphasics, but its ability is compromised when processing demands increase.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.142-144
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• 2002

VoIP(Voice over IP) 시스템을 구현함에 있어서 호설정을 처리하는 여러 프로토콜이 제안되고 있는 가운데 IETF(Internet Engineering Task Force)에서 제안한 SIP(Session Initiation Protocol)는 텍스트 기반의 프로토콜로서 구현과 파싱이 쉬운 등 많은 장점을 가지고 있어 차세대 VoIP의 표준으로 자리잡고 있다. 또한, 뛰어난 확장성을 가지고 있어 다양한 서비스에 적용할 수 있는 호설정 프로토콜이다. 본 논문에서는 SIP를 이용한 VoIP 시스템을 구현함에 있어 주요 구성요소 중 하나인 UA(User Agent)를 2002년 6월에 발표된 새로운 SIP 버전에 맞추어 개발하였다. 본 UA는 플랫폼에 독립적으로 기능을 할 수 있도록 자바(Java)를 사용하여 GUI(Graphical User Interface)환경으로 구현하였다 그리고 RTP(Real-time Transport Protocol) 전송 모듈을 통하여 호설정이 이루어진 후 실제 음성과 화상통신이 이루어지는 부분을 포함하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.295-297
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• 2008

Session Initiation Protocol (SIP) has become the call control protocol of choice for Voice over IP (VoIP) networks because of its open and extensible nature. However, the integrity of call signaling between sites is of utmost importance, and SIP is vulnerable to attackers when left unprotected. Currently a hop-by-hop security model is prevalent, wherein intermediaries forward a request towards the destination user agent server (UAS) without a user agent client (UAC) knowing whether or not the intermediary behaved in a trusted manner. This paper presents an integrated security model for SIP-based VoIP network by combining hop-by-hop security and end-to-end security.

• Journal of Digital Contents Society
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• v.14 no.3
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• pp.357-365
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• 2013

The popularization of smart phones changes the usage patterns of mobile communication from voice-centric to data-centric communication. The demand for wireless data communications is rapidly increasing, and thus the need for expanding infrastructure for mobile communication is also rapidly increasing. In this paper, we propose a scheme for reducing the cost for the mobile communication infrastructure by exploiting opportunistic networks in dissemination of multimedia contents. By using this scheme, the large portion of the cost for mobile communication infrastructure could be saved, and the need of users for multimedia contents could be also fulfilled. Our scheme is evaluated using agent-based simulations. The simulation results show that about 70% of mobile communication can be replaced with the data communication through opportunistic networks.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.13-15
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• 2003

SIP(Session Initiation Protocol)는 사용자간의 멀티미디어 세션을 처리하기 위한 응용 계층의 시그널링 프로토콜로서 유연성 및 확장이 용이한 장점을 가지고 있다. Caller Preference는 이러한 SIP의 기본적인 프로토콜을 확장한 형태로서 송신자가 Preference를 명시하여 서버가 처리할 응답 기능을 선택하거나 수신자의 수신 능력(Callee Capabilities)에 따라 적절한 호처리를 진행할 수 있는 서비스이다. 본 논문에서는 SIP를 기반으로 하는 VoIP(Voice over IP) 시스템을 구현함에 있어 UA(User Agent)내에 Preference를 선택적으로 명시할 수 있는 기능을 포함시키고 또한 이의 요청에 대한 수용이 가능하고 호처리를 진행할 수 있는 네트워크 서버를 구현하였다.