• International Journal of Computer Science & Network Security
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• 제22권2호
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• pp.175-184
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• 2022

Voice over Internet Protocol (VoIP) has grown in popularity as a low-cost, flexible alternative to the classic public switched telephone network (PSTN) that offers advanced digital features. However, additional security vulnerabilities are introduced by the VoIP system's flexibility and the convergence of voice and data networks. These additional challenges add to the normal security challenges that a VoIP system's underlying IP data network infrastructure confront. As a result, the VoIP network adds to the complexity of the security assurance task faced by businesses that use this technology. It's time to start documenting the many security risks that a VoIP infrastructure can face, as well as analyzing the difficulties and solutions that could help guide future efforts in research & development. We discuss and investigate the challenges and requirements of VoIP security in this research. Following a thorough examination of security challenges, we concentrate on VoIP system threats, which are critical for present and future VoIP deployments. Then, towards the end of this paper, some future study directions are suggested. This article intends to guide future scholars and provide them with useful guidance.

• Journal of Internet Computing and Services
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• 제10권6호
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• pp.27-36
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• 2009

PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.

• Proceedings of the KIEE Conference
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• 대한전기학회 1999년도 하계학술대회 논문집 G
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• pp.3249-3252
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• 1999

In this paper, we present the Tele-PACS of radiology, which uses the communication network as asymetric satellite data communication system. The asymetric satellite data communication system uses receive-only satellite links for data delivery and PSTN(Public Switched Telephone Network) modem or N-ISDN(Integrate Services Digital Network) for communication. The satellite communication linking shows the very high-speed performance than terrestial linking such as 28.8 kbps modem linking and 56Kbps linking. The satellite linking is 5 - 10 times faster than the 56Kbps linking. We conclude that 1) Satellite networking is currently the cheapest and fastest solution for internet access. 2) Web-based Image-Viewer enables small size hospitals in rural area to connect to central PACS easily and to retrieve the image data reliably. 3) The suggested teleradiology system using satellite networking could be adequate to fast telemedicine and telecare for rural hospitals especially located in geographically isolated areas such as islands.

• The Journal of the Korea institute of electronic communication sciences
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• 제6권1호
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• pp.55-60
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• 2011

The mobile internet telephony service rapidly grows according to the extending deployment of smartphones. Unlike telephony service over a conventional public switched telephone network (PSTN) or mobile network, internet telephony service cannot guarantee its service quality, which can be severer in a wireless environment. Therefore, a more strict and systematic quality management is required for successful settlement and popularization of mobile internet telephony service. Existing quality management scheme using a specific measurement equipment cannot measure all the time so that it performs late management. In order to overcome the problem, this paper develops a software that can be equipped on a user terminal and measures the service quality all the time. By using the developed software, all-time and user-activating service quality monitoring can be supported.

• Journal of the Korea Society of Computer and Information
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• 제12권1호
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• pp.137-144
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• 2007

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.

• Speech Sciences
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• 제6권
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• pp.163-179
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• 1999

For the set up, management and repair of a mobile communication system, continuous estimation of speech quality is required. Speech quality measurement can be conducted by listener's judgement in a subjective test such as MOS (Mean Opinion Score) test. However, this method is laborious, expensive and time-consuming, it is advisable to predict subjective speech quality via objective measures. This paper presents a robust objective speech quality measure, PLP-CMS (Perceptual Linear Predictive-Cepstral Mean Subtraction), which can predict subjective speech quality in mobile communication systems. PLP-CMS has a high correlation with subjective quality owing to PLP (Perceptual Linear Predictive) analysis and shows a robust performance not being influenced by PSTN (Public Switched Telephone Network) channel effects due to CMS (Cepstral Mean Subtraction). To prove the performance of our proposed algorithm, we carried out subjective and objective quality estimation on speech samples which are variously distorted in a real mobile communication system. As a result, we demonstrated that PLP-CMS has a higher correlation with subjective quality than PSQM (Perceptual Speech Quality Measure) and PLP-CD (Perceptual Linear Predictive-Cepstral Distance).

• Proceedings of the Korean Information Science Society Conference
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• 한국정보과학회 2011년도 한국컴퓨터종합학술대회논문집 Vol.38 No.1(D)
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• pp.215-218
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• 2011

VoIP(Voice over Internet Protocol) 서비스는 기존의 음성전화 서비스(Public Switched Telephone Network, PSTN)와 달리 IP 프로토콜을 이용한 저렴한 통신비용 등의 장점이 있는 음성통신 기술로써, 기존의 아날로그 음성전화 서비스를 대신하는 서비스이며, 새로운 인터넷 융합서비스로 많은 사용자가 이용하고 있다. 하지만 VoIP 서비스가 인터넷망을 이용함으로 IP Spoofing, DoS (Denial of Server) / DDoS(Distributed Denial of Service), 등의 여러 가지 보안의 문제점을 가지고 있다. VoIP 서비스에서 DDoS 공격은 Proxy 서버 등에 대량의 공격 메시지를 보냄으로써 서버의 자원을 고갈시켜 정상적인 서비스를 하지 못하게 한다. DoS, DDoS 공격 중 Invite Flooding 공격은 1분에 수천 개의 Invite 메시지를 보내 회선의 자원을 고갈시키는 공격이다. 특히 IP/Port 위조하여 공격 경우 공격 패킷 탐지하기 어려우므로 차단할 수 없다. 따라서 본 논문에서는 VoIP의 DoS/DDoS 중 하나인 Invite Flooding 공격 시 SIP Proxy Server에서 메시지 분산시키는 방법과 MAC Address와 사용자 번호 등 IP 이외의 고정적인 사용자 정보를 확인하여 공격을 탐지하고, 공격 Agent에 감염된 Phone을 공격차단서비스로 보내 복구시키는 방법을 제안한다.

• Journal of the Institute of Electronics Engineers of Korea TE
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• 제39권4호
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• pp.395-402
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• 2002

In the conventional public switched telephone network(PSTN), there are various types of modulation that can be used in ADSL to offer fast data communication, two of which are CAP(Carrierless Amplitude Phase) and DMT(Discrete MultiTone). As we consider the current situation, DMT is getting more predominant in the market than CAP. One of the reasons is that it gives high performance in spite of its high complexity Since DMT divides the full range of bandwidth into 256 sub-channels, it can be highly adaptive in the circumstances, where the problems of attenuation and noise caused by the propagation distance are very crucial. In this paper, a new bit loading algorithm for DMT modulation is proposed. The proposed algorithm can be efficiently implemented in a way that it requires less computation than the conventional modulation techniques. In contrast to the conventional algorithms which perform sorting processing, the proposed algorithm uses look-up tables to reduce the repetition of calculation. Consequently, it is shown that less processing time and lower complexity can be achieved.

• Journal of the Korean Institute of Telematics and Electronics S
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• 제35S권8호
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• pp.68-76
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• 1998

In this paper, a new morphological image segmentation algorithm of hierarchical structure by binary split is proposed. It splits a region with the lowest quality into two regions using only two markers having the highest contrast. Therefore, it improves the quality of image with limited regions and reduces contour information which is not sensitive to human visual system, when compared with the conventional algorithm. It is appropriate to PSTN, LAN, and mobile networks, of which the available transmission bandwidth is very limited, because the number of regions can be controlled. And the proposed algorithm shows very simple structure because it doesn't need post processing to eliminate small regions and reduces much computation by using only structuring element of small size at simplification step of each hierarchical structure when compared with the conventional algorithm.

• Convergence Security Journal
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• 제8권4호
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• pp.41-49
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• 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.