• Proceedings of the Korea Information Processing Society Conference
• /
• 2000.10b
• /
• pp.1025-1028
• /
• 2000

본 논문에서는 VoIP(Voice over IP)의 광대역 네트워크에서 QoS(Quality of Service)를 지원하기 위한 방안을 스케줄링을 중심으로 하여 연구하였다. 이를 위하여 라우터 중심의 포워딩(forwarding)에 있어서 재생손실(Play Out Loss)이 발생하는 버퍼에 대해서는 음성통신을 제외한 일반 데이터에 할당을 하고 들어오는 음성 데이터에 대해서는 토큰 할당 방식으로 자원을 할당하는 모델에 대해 제안하였다. 또한 음성 데이터에 대해서는 폐기에 대한 방법대신 거절 개념을 포함시켜 이를 EF(Expedited Forwarding)모델과 시뮬레이션을 통하여 분석을 하였다.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.15 no.6
• /
• pp.1256-1264
• /
• 2011

MBS (Multicast Broadcast Service) is defined in WiBro networks for implementing multicast-based services. However, most of the WiBro networks currently used in Korea do not have the MBS functionality and it causes some difficulty in implementing multicast-based services. This paper evaluates the performance of VoIP-based group communication services when implementing using the following two cases: unicast-based and multicast-based group communication systems. The performance evaluation is done using QualNet for each case in terms of the amount of network resource for the service, the delay and delay jitter of packets, and the difference of the delay of members in a group. The simulation result shows that the number of groups and members in a group in a WiBro network is limited because the amount of network resource for the service is increased according to the number of members in a group, and so, the MBS service is required in a WiBro network for a fully-fledged service of VoIP-based group communications. The simulation result also shows that, when a packet bundling is used, the number of groups and members in a group that can be supported in a WiBro network can be increased due to the decrease of the amount of network resource for the service.

• Journal of KIISE:Computing Practices and Letters
• /
• v.14 no.3
• /
• pp.261-265
• /
• 2008

As increasing the user of VoIP service using ITSP(Internet Telephony Service Provider), the VoIP spam becomes a big problem. The spam used in the existing public telephone is detected by using the pattern inspection of call behavior because it is difficult to filter contents for the characteristic of real-time voice communication. However there is a false-positive problem. The threat on spam remains where spam with low threshold can't be detected or users share one number. In this paper, we propose anti-spam for VoIP based on luring test. The proposed method gives a user luring test and he/she can connect to a receiver if passing turing test. A ticket is given to a user that pass luring test and it reduces overhead of luring test in re-dial. The proposed method is implemented on ASUS WL-500G wireless router and Asterisk IP-PBX. Experimental results show the effectiveness of the proposed method.

• Journal of information and communication convergence engineering
• /
• v.6 no.1
• /
• pp.68-72
• /
• 2008

The VoIP(Voice over IP) has been worldwide used and already put to practical use in many fields. However, it is needed to ensure secret of VoIP call in a special situation. It is relatively difficult to eaves-drop the commonly used PSTN in that it is connected with 1:1 circuit. However, it is difficult to ensure the secret of call on Internet because many users can connect to the Internet at the same time. Therefore, this paper suggests a new model of Internet telephone for eavesdrop prevention enabling VoIP(using SIP protocol) to use the VPN protocol and establish the probability of practical use comparing it with Internet telephone.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2014.11a
• /
• pp.218-220
• /
• 2014

VoIP(Voice over Internet Protocol)는 음성 및 화상과 같은 멀티미디어 세션을 인터넷과 같은 IP 기반 네트워크를 통해 통신하는 기술이다. 최근에는 기존의 PC 시스템 이외에 이동통신기기와 다양한 무선네트워크 기반 휴대용 기기들의 보급으로 VoIP 의 사용량은 크게 증가하고 있다. 하지만, 무선네트워크는 그 특성과 환경적 요인으로 NAT 에서의 차단, 지연, 유실등과 같이 통신의 연속성을 보장해 주지 못하는 문제가 발생할 수 있다. 본 논문에서는 무선네트워크에서 통신할 때 발생할 수 있는 이런 문제들에 대응하는 해결 방안을 제시하고 RTP 미디어 재생의 연속성을 보장하는 ICE 기반 다자간 VoIP 시스템 설계와 구현모델에 대해서 기술하고자 한다.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2013.11a
• /
• pp.257-258
• /
• 2013

VoIP(Voice over Internet Protocol)는 인터넷과 같은 IP 망에서 음성과 영상을 전송하기 위한 기술이며, 차세대 망에서의 음성, 테이터 통신을 위한 기술로 부상되고 있다. 따라서 VoIP 응용은 인터넷 망을 이용하는 다양한 단말기들 사이의 음성 및 영상 통신을 위하여 더욱더 많이 사용되어 질 것으로 예상된다. 본 논문에서는 VoIP 기술을 여러 분야에 적용할 수 있는 응용성과 실제 다자간 음성통신의 구현 방법에 대해서 기술하겠다.

• International journal of advanced smart convergence
• /
• v.9 no.4
• /
• pp.209-214
• /
• 2020

VoIP technology has been widely used for exchanging voice or image data through IP networks. VoIP technology, often called Internet Telephony, sends and receives voice data over the RTP protocol during the session. However, there is an exposition risk in the voice data in VoIP using the RTP protocol, where the RTP protocol does not have a specification for encryption of the original data. We implement programs that can extract meaningful information from the user's dialogue. The meaningful information means the information that the program user wants to obtain. In order to do that, our implementation has two parts. One is the client part, which inputs the keyword of the information that the user wants to obtain, and the other is the server part, which sniffs and performs the speech recognition process. We use the Google Speech API from Google Cloud, which uses machine learning in the speech recognition process. Finally, we discuss the usability and the limitations of the implementation with the example.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2010.10a
• /
• pp.153-155
• /
• 2010

Internet phone (VoIP) services in Korea have achieved noticeable growth year after year since the service launching, and the growth still coninues. The market of mobile internet phone also expands sharply. Therefore, it is crucial to deploy networks which can support mobile internet phone services with excellent quality. For mobile internet phone services, it will be necessary to build and use networks with good mobility and high transmission rate. Current wireless networks for Internet services include 3G, Wi-Fi, and mobile WiMAX networks. 3G provides good mobility but lower transmission rate, whereas Wi-Fi exhibits excellent transmission rate but less mobility. Mobile WiMAX networks taking the merits of both, high mobility and transmission rate, are being deployed widely in recent years. This article examines the recent standardization efforts of WiMAX Forum for VoIP service in WiMAX networks.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.50 no.7
• /
• pp.10-18
• /
• 2013

With advancing of mobile communication services and commercializing of VoLTE (Voice of LTE), it is getting to pay attention on QoS of VoLTE. This paper proposes an integrated E-model in which some factors influenced to service quality of VoIP and VoLTE based voice communication system are considered in calculating the voice quality of Wideband Codec. The model aims to calculate R value which reflects the situations of access network, network characteristics, terminals' usage and mobility. We mainly deal with the integrated E-model's structure, related algorithms and optimal parameters for VoLTE. Some experiments show that the voice quality difference between VoIP and VoiceChecker, and VoLTE and POLQA, is below 10%. With the proposed model, we can calculate the voice quality by making use of the factors directly affected to service quality and the environment of VoLTE terminal and network. As a result, we can estimate the service quality in advance, without measuring it in real wireless environment.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2013.11a
• /
• pp.943-944
• /
• 2013

모바일 VoIP 음성통화는 지연, 지터 그리고 패킷 손실과 같은 네트워크 장애요소로 인해 품질저하가 발생된다. 본 논문에서는 이러한 네트워크 장애요소로 인한 음성통화 품질이 저하되는 문제를 해결하기 위하여 저런력 및 고품질 음성통화를 위한 지능형 모바일 VoIP 시스템을 제안한다. 제안된 방식은 손실은틱, 지터추정, 플레이아웃 스케줄링 등의 모듈로 구성되어 있다. 성능 측정 결과, 제안된 알고리즘은 기존 알고리즘에 비해 높은 PESQ와 낮은 버퍼링 지연을 보여주었다.