• Journal of Digital Convergence
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• v.9 no.5
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• pp.225-233
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• 2011

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented a recording system which records voice data of SIP-based Internet telephone's voice calls. It is developed on the linux system and has some features such as audio mixing of two in/out voice channels, live packet sniffing, and the ability to transfer mixed audio files to the log file server. These functions are implemented using various open source softwares. Afterwards, this VoIP recording system will be applied as a base technology to advanced services like a VoIP-based call center system.

• 한국HCI학회:학술대회논문집
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• 2007.02b
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• pp.85-90
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• 2007

세계 이동 통신 시장은 점차 3G(세대)~3.5G(세대)로 전환되면서 세대 교체하고 있다. 3G 이동 통신의 대표 서비스인 화상전화는 음성통화와는 다른 새로운 커뮤니케이션이나 여러 가지 원인으로 일반화되지 않고 있는데, 그 원인 중 하나가 새로운 사용 경험과 사용 환경이 충분히 고려되지 않은 UI 라고 생각된다. 따라서, 본 조사에서는 한국, 일본, 영국, 이탈리아 등 4 개국을 대상으로 현재 출시된 화상전화 제품에 대하여 실사용자 FGI 와 설문, In-depth interview 등을 통해 사용자들의 화상전화 사용 경험을 조사하고, 조사를 통한 발견점을 통해 UI 설계 전문가 관점에서 향후 UI 에 대한 제안점을 찾아 보다 나은 화상전화 경험을 제공하기 위한 바탕으로 삼고자 한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.11
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• pp.2498-2511
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• 1997

Due to the recent advances in computer technologies and high-speed networks, telecommunication services and provision of multimedia services will be considered a new software architecture adapting networking infrastructure on information networking architecture. In this paper, the video conference services has been selected as a target service example because it is expected to become one of the most important services on the full service network. In fact, it can be viewed as the basis for providing telecommunication services such as video telephony and video conferencing. This paepr presents the prototyping of TINA(Telecommunication Information Networking Architecture) based desktop video conference system using the concepts of session management. The prototyping of deskop video conference system aims at assessing TINA concepts and refinement of the mapping between session graph and connection graph, and provides support of open service platform towards distribution and objected-orientation.

• The Transactions of the Korea Information Processing Society
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• v.4 no.12
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• pp.3097-3110
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• 1997

As to the recent advances in computer technologies and high-speed networks, telecommunication services and provision of multimedia services will be provided on the basic of a new software architecture adapting networking infrastructure on open information networking architecture. In this paper, the video conference services has been selected as a target service example because it is expected to become one of the most important services on the full service network. In fact, it can be viewed as the basis for providing telecommunication services such as video telephony and video conferencing. This paper presents the prototyping of TINA (Telecommunication Information Networking Architecture) based desktop video conference system using the concepts of session management. The prototyping of desktop video conference system aims at assessing TINA concepts and refinement of the mapping between session graph and connection graph, and provides object modeling methodologies towards distribution and objected-orientation.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.17-24
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• 2005

Traditional Internet applications such as FIP and E-mail are increasingly sharing bandwidth with newer, more demanding applications such as Web browsing, IP telephony, video conference and online games. These new applications require Quality of Service (QoS), in terms of delay, loss and throughput that are different from QoS requirements of traditional applications. Unfortunately, current Active Queue Management (AQM) approaches offer monolithic best-effort service to all Internet applications regardless of the current QoS requirements. This paper proposes and evaluates a new AQM technique, called MCDT that provides dynamic and separated buffer threshold for each Applications, those are FTP and e-mail on TCP traffic, streaming services on tagged UDP traffic, and the other services on untagged UDP traffic. Using a new QoS metric, our simulations demonstrate that MCDT yields higher QoS in terms of the delay variation and a packet loss than RED when there are heavy UDP traffics that include streaming applications and data applications. MCDT fits the current best-effort Internet environment without high complexity.

• Journal of the Korea Society of Computer and Information
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• v.4 no.2
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• pp.46-60
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• 1999

Recently, CTI (Computer Telephony Integration) technology has been widely applied to various area such as video conference, file transfer, voice mail, automatic message transfer and automatic redial, integrated messaging and network fax. In this paper, an administrative affairs supporting system using call control mode of CTI is designed. To improve inefficient processing of job due to heavy calling from entrance candidates during entrance examination of a college, the system is developed. The database of the system is desigend using object modeling technique. Also, the automatic calling and response system using CTI call control mode is implemented. Especially, to interface with voice of candidates who ask whether they pass or fail the entrance examination of the college, TTS(Text To Speech) module is developed.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.157-159
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• 2003

본 논문에서는 화상 통화를 위한 비디오 인코딩에서 화자의 얼굴 화질을 강화하여 인코딩 할 수 있는 동영상 인코딩 방법을 제안한다. 제안한 인코딩 방법에서는 이미지의 Cr 데이터 및 움직임벡터 정보를 이용하여 빠르게 화자 얼굴 영역을 검출하고, 이 영역에 대하여 선택적인 양자화를 통하여 상대적으로 많은 비트량을 할당하여 화자의 얼굴 화질을 상대적으로 강화한다. 이 방법을 H.263 인코더에 적용하는 경우 전체적으로 이런 방법을 적용하지 않았을 때와 비교하여 18% 정도의 추가적인 CPU 오버헤드가 필요하였지만, 얼굴 영역에 대하여서는 PSNR 3dB 정도의 화질이 개선될 수 있음을 실험을 통하여 증명하였다.

• Journal of Communications and Networks
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• v.8 no.4
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• pp.432-441
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• 2006

As the community antenna television (CATV) networks becomes ubiquitous, instead of constructing an entirely new broadband network infrastructure, it has emerged as one of the rapid and economic technologies to interconnecting heterogeneous network to provide broadband access to subscribers. How to support ubiquitous real-time multimedia applications, especially in a heavy traffic environment, becomes a critical issue in modern CATV networks. In this paper, we propose a time guaranteed and efficient upstream minislots allocation algorithm for supporting quality-of-service (QoS) traffic over data over cable service interface specification (DOCSIS) CATV networks to fulfill the needs of realtime interactive services, such as video telephony, video on demand (VOD), distance learning, and so on. The proposed adaptive fast expansion algorithm and the loading statistics with dynamic swapping algorithm have been shown to perform better than that of the multimedia cable network system (MCNS) DOCSIS.

• Proceedings of the Korea Information Processing Society Conference
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• 2009.11a
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• pp.375-376
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• 2009

모바일 환경 상에서 영상통화는 제한된 대역폭 및 손떨림 현상 등의 문제로 인하여 저품질의 서비스가 제공되고 있다. 기존의 영상통화 환경에서 향상된 품질의 영상통화 서비스를 제공하기 위하여 중요 객체와 비중요 객체를 구분하여 분리하며, 중요 객체에 대해서는 고화질을 갖도록 하는 비디오 압축 방법을 제안한다. 매 프레임 마다 중요 객체와 비중요 객체를 분리하며, 분리된 객체는 다른 비트율과 화질을 갖도록 각 객체에 적합한 변환 커널을 적용하여 부호화 절차를 달리한다. 중요 객체는 저압축, 고화질을 가지도록 부호화하며, 비중요 객체에 대해서는 고압축, 저화질을 가지도록 부호화함으로써 제한된 대역폭 내에서 비트율을 유지하도록 한다.

• Journal of Communications and Networks
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• v.9 no.4
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• pp.446-456
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• 2007

Increased access to broadband networks has led to a fast-growing demand for voice and video over IP(VVoIP) applications such as Internet telephony(VoIP), videoconferencing, and IP television(IPTV). For pro-active troubleshooting of VVoIP performance bottlenecks that manifest to end-users as performance impairments such as video frame freezing and voice dropouts, network operators cannot rely on actual end-users to report their subjective quality of experience(QoE). Hence, automated and objective techniques that provide real-time or online VVoIP QoE estimates are vital. Objective techniques developed to-date estimate VVoIP QoE by performing frame-to-frame peak-signal-to-noise ratio(PSNR) comparisons of the original video sequence and the reconstructed video sequence obtained from the sender-side and receiver-side, respectively. Since processing such video sequences is time consuming and computationally intensive, existing objective techniques cannot provide online VVoIP QoE. In this paper, we present a novel framework that can provide online estimates of VVoIP QoE on network paths without end-user involvement and without requiring any video sequences. The framework features the "GAP-model", which is an offline model of QoE expressed as a function of measurable network factors such as bandwidth, delay, jitter, and loss. Using the GAP-model, our online framework can produce VVoIP QoE estimates in terms of "Good", "Acceptable", or "Poor"(GAP) grades of perceptual quality solely from the online measured network conditions.