Earticle

텔레메틱스 기반의 VoIP 음성통화는 지연, 지터 그리고 패킷손실과 같은 네트워크 장애요소로 인해 품질저하가 발생된다. 본 논문에서는 이런 장애요소로 인한 통화품질 문제를 해결하기 위하여 수신단 기반의 통화품질 향상 알고리즘을 제안한다. 제안된 방식은 손실은닉, 역동적인 지터추정을 이용한 적응적인 플레이아웃 버퍼 스케줄링, 그리고 전이영역에 존재하는 두 신호간의 스무딩 방법으로 구성된 플레이아웃 콘트롤 및 신호합성부를 통해 고품질의 음성통화를 가능하게 한다. 성능 측정 결과, 제안된 알고리즘은 기존 알고리즘에 비해 높은 PESQ와 낮은 버퍼링 지연을 보여주었다.

The quality of real-time voice communication over Internet Protocol networks based on telematics is affected by network impairments such as delays, jitters, and packet loss. To resolve this issue, this paper proposes a receiver-based enhancing method of VoIP speech quality. The proposed method enables users to deliver high-quality voice using playout control and signal reconstruction, which consists of concealment of lost packets, adaptive playout-buffer scheduling using active jitter estimation, and smooth interpolation between two signals in a transition region. The proposed algorithm achieves higher Perceptual Evaluation of Speech Quality (PESQ) values and low buffering delay than the reference algorithm.

Which factors should influence consumer consideration to subscribe to Voice over Internet Protocol (VoIP)? Policy issues, managerial concerns, and demographic variables are possible factors. This paper discusses policy and managerial issues regarding VoIP adoption. A model that explains VoIP adoption is proposed and tested. This study analyzes a survey of 750 prospective VoIP users in Korea. The testing is accompanied by logistic regression and discriminant analysis. The results show that trust in VoIP, relative comparison of quality to fixed service, numbering plan, satisfactions of call quality and customer services on both fixed and mobile services have impacts on the adoption of VoIP. Implications for VoIP providers and policy makers are presented.

정부는 2004년 IT839 전략의 하나로 VoIP를 8대 신규 서비스로 지정하였고, 정부와 일부 사업자에 의해 적극적으로 추진되었다. 인터넷전화는 유선전화와는 달리 이동성이 보장되어 어디서나 인터넷 단말기만 가지고 있으면 인터넷에 연결하여 통화를 할 수 있을 뿐만 아니라 각종의 부가서비스(영상통화, 데이터전송, 인터넷접속 등)가 가능하다는 장점과 더불어 이용요금도 이동전화와 유선전화와 비교했을 때 저렴한 편이다. 국내 통신 기업들은 포화된 국내 통신서비스 시장의 새로운 비즈니스모델 창출을 위해 각종의 부가서비스 제공이 가능한 인터넷전화에 주목하고 있고 통신 소비자들은 저렴한 인터넷 전화요금과 우리나라의 가계당 통신비중이 높다는 점, 그리고 각종 부가서비스가 제공되는 인터넷전화의 긍정적 요인에 크게 반응할 것으로 보인다. 현재는 VoIP가 유선전화시장을 급속하게 대체할 것으로 예측되며, 초고속 하향 패킷접속과 와이브로(WiBro: Wireless Broadband Internet)기반의 이동통신에서 VoIP가 장착이 된다면 이동전화를 위협하는 Killer Application으로 자리 잡게 될 것으로 보인다. 또한, 이 기술은 단지 음성전화 뿐만 아니라 SMS(Short Message Service), 음성채팅, 영상통화 등의 다양한 부가 서비스가 가능하며 통신․방송의 융합에 따른 방송서비스까지 가능한 실시간 멀티미디어 서비스 제공을 위한 핵심기술이라는 점에서 최근에 더욱 관심이 증폭되고 있다. 결국 인터넷 전화는 기존의 일반전화가 가진 장단점을 모두 흡수하여 새로운 혁신을 가져 올 것으로 예상된다. 향후 인터넷전화의 발전과 대중화를 위한 발판을 마련하기 위한 후속적 연구가 지속되어야 할것이다.

Internet was first designed only for sending data, but as the time passed,internet started to evolve into a broadband multi-media web that is capable of transmitting sound, video, high-capacity data and more due to the demands of internet users and the rapid changing internet-communication technology. Domestically, in January, 2000 Saerom C&T, launched a free VoIP, but due to limited ways of conversation(PC to PC) and absence of a revenue model, and bad speech quality, it had hit it's growth limit. This research studied VoIP based on technological enhancement in super-speed internet. According to IDC, domestic internet market's size was 80,800 million in 2008, and it formed a percentage of 12.5% out of the whole soundcommunication market. in case of VoIP, it is able to maximize it's profit by connecting cable and wireless network, also it has a chance of becoming firm-concentrated monopoly market by fusing with IPTV. Considering the fact that our country is insignificant in MVNO revitalization, regulating organizations will play a significant roll on regulating profit between large and small businesses. Further research should be done to give VoIP a secure footing to prosper and become popularized.

In Voice over Internet Protocol (VoIP) system, the speech signal is degraded when passed through the network layers. The speech signal is processed through the best effort policy based IP network, which leads to the network degradations including delay, packet loss and jitter. The packet loss is the major issue of the degradation in the VoIP signal quality; even a single lost packet may generate audible distortion in the decoded speech signal. In addition to these network degradations, the quality of the speech signal is also affected by the environmental noises and coder distortions. The signal quality of the VoIP system is improved through the interleaving technique. The performance of the system is evaluated for various types of noises at different network conditions. The performance of the enhanced VoIP signal is evaluated using wideband extension-perceptual evaluation of speech quality (WB-PESQ) measurement for wideband signal.

Voice-over-IP (VoIP) is being widely used to packetize and relay voice information over existing IP networks. Due to the cheaper costs involved in using VoIP as opposed to traditional PSTN networks it is quickly gaining momentum and has seen widespread use in both personal and business domains. If this technology is to grow and gain over normal phone calls it must fulfill certain conditions that Public Switched Telephone Networks (PSTN) currently provide. Since VoIP uses the same routes used by normal Internet traffic, it is prone to a wide range of security threats, similar to those faced by normal packet data. These include, but are not limited to, Denial of Service (DoS), intruders in the network who either eavesdrop or are able to make free calls, man-in-the-middle attacks, etc. It thus becomes imperative that a protocol be developed that is able to both guard against and take corrective action in the event of the occurrence of any attack. In this paper we propose a protocol that detects and prevents the occurrence of a distributed Denial of Service (DDoS) attack. It also enforces security mechanisms to ensure that an attacker has not gained access to a user's password and thus making illegitimate calls. We develop a trust mechanism that can be used to detect an attacker in the network. The protocol has been rigorously examined through a wide range of experiments and the results have been found to be quite promising.

유무선 통신 기술, 디지털 미디어, 그리고 영상처리 기술의 비약적인 발전과 서비스 통합화 추세는 광대역통합망(BcN: Broadband convergence Network)과 같은 초고속 네트워크를 통하여 VoIP, IPTV와 같은 여러 형태의 새로운 서비스를 창출하게 되었다. VoIP 서비스가 기존의 공중회선 교환망에서와 같은 이윤 창출을 위해서는 기존 서비스 이상의 품질을 제공하여야 한다. 따라서 실시간 품질측정 프레임워크는 VoIP 서비스를 제공하기 위한 가장 중요한 요소라고 할 수 있다. 이를 위해 IETF (Internet Engineering Task Force)에서는 RTCP (Real-Time Transport Protocol Control Protocol)를 확장한 RTCP-Extended Reports (RTCP-XR)을 정의하였다. 그러나 RTCP-XR에서는 음성품질을 측정하기 위한 항목만을 정의하였을 뿐 실제 VoIP 품질 측정을 위한 절차와 방법은 규정하지는 않았다. 본 논문은 종단간 패킷화된 음성을 효과적으로 측정하기 위한 프레임워크 제시를 목적으로 하고 있다. 이를 위해 본 논문에서는 VoIP 품질 측정개념과 더불어 제안된 프레임워크에서 측정방법을 단계적으로 기술하였다. 아울러 RTCP-XR의 개념을 확장한 새로운 형태의 포맷을 제안하였다.

Advances in networking technology digital media, and codecs have made it possible for the Internet evolves into a Broadband convergence Network (BcN) and provides various services including Voice over Internet Protocol (VoIP) and IPTV over their high-speed IP networks. In order for the Internet to make a profit as traditional Public Switched Telephone Network (PSTN), it must provide high qualify VoIP services. Therefore, real time qualify measurement framework is the most important requisite to provide VoIP service. For this, IETF (Internet Engineering Task Force) defined RTCP-Extended Reports (RTCP-XR) that extend RTCP (Real-Time Transport Protocol Control Protocol). However, procedure and method tot actually VoIP qualify measurement did not recommended nothing but defined item to measure voice quality. Our objective in this paper is to describes a practical measuring framework for end-to-end QoS of switched voice packet in an IP environment. It includes concepts as well as step-by-step procedures for measuring packetized voice streams. It also proposes new formats that extend RTCP-XR's concept.

This study addresses the issue regarding the difference of user perception on Voice over Internet Protocol(VoIP) for the users with different cultural background. Using the variables identified by literatures, this study tries to find the variables which causes the users’ satisfaction for VoIP. Based on the cultural characteristics on VoIP users, a conceptual framework of VoIP users’ satisfaction was also developed to investigate the perception of VoIP users. The results of the study are expected to show how some of verified variables would affect the user’s perception for VoIP platform.

Which factors should influence consumer consideration to subscribe to Voice over Internet Protocol (VoIP)? Policy issues, managerial concerns, and demographic variables are possible factors. This paper discusses policy and managerial issues regarding VoIP adoption. A model that explains VoIP adoption is proposed and tested. This study analyzes a survey of 750 prospective VoIP users in Korea. The testing is accompanied by logistic regression and discriminant analysis. The results show that trust in VoIP, relative comparison of Quality to fixed service, numbering plan, satisfactions of call Quality and customer services on both fixed and mobile services have impacts on the adoption of VoIP. Implications for VoIP providers and policy makers are presented.

Voice over Internet Protocol (VoIP) has received a great deal of attention over the past several years. The growing popularity of VoIP has opened up new opportunities for saving money not only for the ordinary household consumers but also for the businesses looking for reducing cost for their daily operation. To examine the effect of cultural differences on the perception for VoIP technology, VoIP users from two markedly different countries in their cultural background. The findings of this study can provide useful insights for both the academic and practitioner community.

행정기관 인터넷전화를 구축하여 사용하는 국가정보통신서비스 'C'그룹 사업자에는 KT, SK브로드밴드, LG유플러스, 삼성SDS 4개 사업자가 있다. 행정기관 인터넷전화에 대한 공격을 대비하여 보안 대응 모델에 대한 연구가 필요하다. 본 논문에서는 행정기관 인터넷전화 사업자 별로 보안 대응 대책을 내용으로 조사 분석한다. 행정기관 인터넷전화 보안 침해 모델 개발 연구를 위해 국정원의 5가지 보안 위협에 대해 기밀성을 중심으로 우선순위를 설정하여 (1)불법 도청 (2)호 가로채기 (3)서비스 오용 (4)서비스거부 공격 (5)인터넷전화 스팸 공격에 대한 공격 시나리오 작성하여 분석한다. 행정기관 인터넷전화 보안 대응 모델 개발 연구는 보안 위협별 보호 기술 분석과 보안 침해 단계별 대응 체계 연구를 통해 단계별 대응 모델 개발 연구를 한다.

Voice over Internet Protocol calls using administrative agency to build a national information and communication service, 'C' group, providers, the KT, SK Broadband, LG U+, Samsung SDS, as there are four operators. To prepare for an attack on Voice over Internet Protocol for administrative agency, security is a need for research to support the model. In this paper, the Internet telephone business of Administrative Agency to investigate and analyze the specific security measures to respond. Should set priorities around confidentiality about five security threats from NIS to Study of Voice over Internet Protocol Security Response Model for Administrative Agency. (1) Illegal wiretapping, (2) call interception, (3) service misuse, (4) denial of service attacks, (5) spam attacks, write about and analyze attack scenarios. In this paper, an analysis of protection by security threats and security breaches through a step-by-step system to address the research study is a step-by-step development of the corresponding model.

스마트폰은 시간과 장소와 기기의 제한을 받지 않는 유비쿼터스 사회로의 전환과 업무에 도입되고 있다. 급속한 스마트폰의 사용 증가는 모바일 업무의 활성화와 행정기관에서도 스마트사회로 전환을 가져왔다. 최근 스마트폰을 이용한 VoIP서비스가 활성화 되고 있지만, 스마트폰과 VoIP가 가지는 무선인터넷 취약점에 노출되고 있다. 본 논문에서는 스마트폰을 이용한 VoIP서비스에 대한 보안성을 강화하기 위한 암호화 기술을 연구한다. 내 외부 신호 및 통화 암호화와 행정기관 인터넷전화 보안 규격을 연구한다. 스마트폰 VoIP서비스에 대한 기기 인증서 보안과 내부 신호 및 통화 암호화를 연구한다. 본 논문은 스마트 시대에 스마트폰 VoIP사용의 안전성과 사용성에 기여할 것이다.

Smart phone is being used in the job as the ubiquitous society will Without being restricted by the time and place and devices. The rapid increase in the use of smart phones has brought the activation of the mobile job. And government agencies have brought in the transition to a smart society. In this paper, using a Voice over Internet protocol(VoIP) service for your smart phones to enhance security is the study of encryption technologies. External and internal signals, and call encryption and security standards of administrative agencies is the study of VoIP. Smart phone VoIP service is a study that security of equipment certificate, the internal signal and call encryption. This paper will contribute what using smart phone VoIP security and usability In smart generation.

최근 인터넷전화는 IP기반에서 유선과 무선을 이용하여 음성뿐만 아니라 멀티미디어 정보전송을 제공한다. 유무선 인터넷전화는 네트워크상에서 인터넷전화 호 제어 신호 및 통화 내용의 불법 도청이 용이하며, 서비스 오용 공격, 서비스 거부 공격의 대상이 될 수 있는 등 기존 유선전화에 비해 여러가지 보안취약점이 존재한다. 본 논문에서는 인터넷전화 단말에서 IP Phone 정보를 획득하기 위한 스캐닝을 통해 IP Phone의 패스워드를 확인하고, 관리자 페이지를 통해 로그인을 성공한다. 그 후 IP Phone 관리자 페이지를 접속한 후 VoIP 설정 화면으로 전화번호, 포트번호, 인증번호 등을 수정한다. 또한 관리자 페이지에서 등록되어 있는 IP Phone들의 통화기록을 확인하여 개인정보 해킹을 연구한다.

Recently, Voice over Internet protocol(VoIP) in IP-based wired and wireless voice, as well as by providing multimedia information transfer. Wired and wireless VoIP is easy on illegal eavesdropping of phone calls and VoIP call control signals on the network. In addition, service misuse attacks, denial of service attacks can be targeted as compared to traditional landline phones, there are several security vulnerabilities. In this paper, VoIP equipment in order to obtain information on the IP Phone is scanning. And check the password of IP Phone, and log in successful from the administrator's page. Then after reaching the page VoIP IP Phone Administrator Settings screen, phone number, port number, certification number, is changed. In addition, IP Phones that are registered in the administrator page of the call records check and personal information is the study of hacking.