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

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.4 s.346
• /
• pp.1-12
• /
• 2006

In this paper we propose an efficient mobile multimedia communication protocol, mobile terminal software platform and mobile VoIP application for IP-based convergence networks. The Proposed mobile multimedia communication protocol is called as ST-MRSVP (Split tunnel based Mobile Resource reServation Protocol) which integrates split tunnel based Mobile IP and RSVP in order to support hish speed mobility. Since mobile terminal platform supports QoS (Qualify of Service) with keeping seamless mobility, mobile QoS supporting modules are developed and interworked together by means of shared memory mechanism. Testbed is composed of a core-network embedding the proposed protocols and wireless LAN-based access networks. We verify functionality and performance of the proposed techniques by using various mobility test over the testbed. As a result, the proposed architecture can reduce the handover delay time with QoS support under 30% comparing with the standard mechanisms and support voice quality as good as CDMA phone.

• Journal of Advanced Navigation Technology
• /
• v.21 no.1
• /
• pp.113-118
• /
• 2017

As interest in mobile devices and networks has increased recently, voice over internet protocol (VoIP) service, which is a technology for transmitting voice data using an existing internet protocol (IP) network, has rapidly spread, Cheap voice call service has become possible. As the digital broadcasting service becomes popular, hybrid fiber coaxial (HFC) network technology, which uses broadband cable network through fusion of broadcasting and communication, utilizes existing communication system and network equipment to provide various new services such as interactive broadcasting service. Therefore, if UGS-AD is applied to VoCM and RTPS is applied to MTA in order to guarantee the quality of voice data in actual HFC Internet service network, it is possible to smoothly perform voice data transmission in narrow upstream band which is a problem in actual commercial HFC network We also proposed a method to improve VoIP service by improving QoS of voice data in HFC Internet service network.

• Proceedings of the IEEK Conference
• /
• 2000.09a
• /
• pp.19-22
• /
• 2000

최근 이동전화의 사용이 급격히 확산됨에 따라 핸즈프리 단말기를 이용한 전화통신의 필요성이 대두되고 있다. 차량내 핸즈프리 통신상황의 경우 근거리에 위치한 스피커와 마이크로폰의 커플링에 의해 발생하는 음향반향과 차량내에 존재하는 배경잡음은 통화 품질을 크게 저하시킨다. 본 논문에서는 이동통신에 적합한 음향반향제거기와 잡음제거기의 결합시스템을 제안하고, 이를 고정 소수점 DSP를 이용하여 실시간 구현하였다. 실시간 구현을 위하여 음향반향제거기에는 NLMS 알고리즘에 의해 구동되는 제한된 차수의 적응반향제거기법을 사용하였다. 잔여반향 및 배경잡음제거를 위해 CDMA방식의 셀룰라 이동통신에 사용되는IS-127 EVRC음성 부호화기의 표준안에 포함된 잡음제거방식을 사용하였다. 제안된 시스템을 16 비트 고정소수점DSP인 OAK DSP Core를 이용하여 약 18.6MIPS의 연산량으로 실시간 구현되었다.

• Electronics and Telecommunications Trends
• /
• v.18 no.5 s.83
• /
• pp.1-10
• /
• 2003

최근의 폭발적인 인터넷 사용의 증가와 또 그것과 유사한 품질의 서비스를 이동중에도 제공 받고자 하는 사용자의 요구와 음성통화 시장의 포화에 따라 무선 데이터 시장을 새로운 돌파구로 보는 제조업체 및 서비스 제공자의 노력으로 IMT-2000 서비스가 이미 진행되고 있다. 비록 2세대 디지털 셀룰러에서 IMT-2000으로의 진화가 현재 지지부진 하지만, 2010년경의 데이터 사용량 예측을 참고로 할 때 새로운 시스템의 출현이 필연적이다. 본 논문에서는 최근까지 논의된 4세대 이동통신에 대한 시스템 및 서비스의 비전을 제시하고자 한다. IMT-2000에서 화상전화를 목표로 하였다면, 4세대에서는 Telepresence의 완벽한 구현과 유비쿼터스 시대로의 첫걸음으로서 다양한 네트워크와 컨버전스를 강조한다. 본 논문에서 이 두 가지 특징을 중심으로 4세대의 비전을 기술하고자 한다.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.48-51
• /
• 1999

본 연구에서는 PCS 이동전화의 객관적 통화품질평가 척도개발을 위한 기초연구로 기존의 CD(Cepstral Distance), MSD (Mel Spectral Distance), BSD(Bark Spectral Distance), PSQM (Perceptual Speech Quality Measure) 척도를 적용하여 그 성능을 비교 분석하였다. 이 척도들을 실제환경에서 수집된 PCS 음성데이터에 대해서 적용하였고 이 결과치와 청취자들의 평가 반응에 의해 얻어진 MOS 결과치와의 상관성을 조사하였다. 실험 결과, BSD와 PSQM 척도의 상관성이 0.81, 0.84로 나타나 CD, MSD보다 성능이 더 우수함을 보였다.

• Electronics and Telecommunications Trends
• /
• v.18 no.1 s.79
• /
• pp.42-50
• /
• 2003

최근의 폭발적인 인터넷 사용의 증가와 또 그것과 유사한 품질의 서비스를 이동중에도 제공 받고자 하는 사용자의 요구와 음성통화 시장의 포화에 따라 무선 데이터 시장을 새로운 돌파구로 보는 제조업체 및 서비스 제공자의 노력으로 IMT-2000 서비스가 이미 진행되고 있다. 비록 2세대 디지털 셀룰러에서 IMT-2000으로의 진화가 현재 지지부진하지만, 2010년경의 데이터 사용량 예측을 참고로 할 때 새로운 시스템의 출현이 필연적이다. ITU-R SG8에서는 1999년 말까지 초기 IMT-2000 시스템에 대한 표준화 작업을 수행해왔던 TG8/1을 해체하고 2000년 이후의 위성 및 지상 부분 모두를 포함한 IMT-2000의 발전과IMT-2000 이후 시스템에 대한 비전과 목적의 정립을 위해 SG8 산하에 WP8F를 조직하여 활발하게 연구하고 있다. 본 고에서는 IMT-2000의 비전 표준화를 중심으로 하여 WP8F의 주요 표준화 동향을 살펴본다. 아울러 WP8F의 유관 기관의 활동도 함께 기술한다.

• Convergence Security Journal
• /
• v.8 no.4
• /
• pp.41-49
• /
• 2008

VoIP is a service that changes the analogue audio signal into a digital signal and then transfers the audio information to the users after configuring it as a packet; and it has an advantage of lower price than the existing voice call service and better extensibility. However, VoIP service has a system structure that, compared to the existing PSTN (Public Switched Telephone Network), has poor call quality and is vulnerable in the security aspect. To make up these problems, TLS service was introduced to enhance the security. In practical system, however, since QoS problem occurs, it is necessary to develop the VoIP security system that can satisfy QoS at the same time in the security aspect. In this paper, a user authentication VoIP system that can provide a service according to the security and the user through providing a differential service according to the approach of the users by adding AA server at the step of configuring the existing VoIP session is suggested. It was found that the proposed system of this study provides a quicker QoS than the TLS-added system at a similar level of security. Also, it is able to provide a variety of additional services by the different users.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.7 no.2
• /
• pp.243-248
• /
• 2012

In this paper we simulate the capacity of VOIP calls through WiFi network by computer simulations using OPNET modeler. The results show that sudden quality degradations occur on all VoIP calls when the number of call of an AP(Access Point) increases beyond a specific value. The reason of the quality degradation was turned out to be the queueing delay at the down link of AP. Under the IEEE 802.11b environments, the maximum number of VoIP calls of an AP maintaining the required voice quality (MOS > 2.5), was evaluated as 5, 12, and 27 when we use G.711, G.729a, and G.729a VAD codec, respectively.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.16 no.6
• /
• pp.9-14
• /
• 2016

The Korea Communication Commission(KCC) measures communication qualities of mobile phones' voice calls, mobile internet, and broadband internet to provide quality communication services. It particularly focuses on measuring qualities of mobile internet data that is on the rise in terms of usage. However, existing quality measurement simply measures the speed of mobile communication through data without providing much information to users and does not disclose measured data, making it hard for individuals to objectively determine their communication quality. Thus, to improve such problems, the author suggests a system that measures qualities of mobile communication through Android. The method guarantees user convenience and reliability and improved mobile communication by making it easy for users to select the quantity of data to be used. It also saves the current GPS obtained through GPS module or mobile communication network and transfers measured data to servers so that the data can be processed into new information and be reinterpreted.