• Journal of Advanced Marine Engineering and Technology
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• v.27 no.1
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• pp.65-75
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• 2003

The internet telephony service is one of the successful internet application services. VoIP is the key technology for the service to come true. VoIP uses H.323 or SIP as the standard protocol for the distributed multimedia services over the internet environment, in which QoS is not guaranteed. VoIP carries the packetized voice by using the RTP/UDP/IP protocol stack. The UDP-based internet services cause the data transmission problem to the users behind the internet firewall. So does the internet telephony service. The users are not able to listen the voices of the counter-parts on the public internet or PSTN. It makes the problem more difficult that the internet telephony service addressed in this paper uses only one UDP port number to send the voice data of all sessions from gateway to terminal node. In this paper, two schemes including the usage of dummy UDP datagrams, and the protocol conversion are suggested. The implementation of one of the schemes, the protocol conversion, and the performance evaluation are described in detail.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.265-268
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• 2011

Voice over Internet protocol(VoIP) is support a VoIP service as All-IP method and Gateway method etc. All-IP method to the unit in an Internet environment by applying both the IP service is an VoIP system. Gateway method, using a normal phone call in a way that the Internet is using VoIP. In this paper, scanning and analyze the vulnerability for VoIP systems and networks from All-IP method and the Gateway method In the test bed. All-IP method and Gateway method found in the VoIP vulnerabilities, hacking attack, a denial of service attacks and VoIP spam attacks are carried out. Through analysis of post-attack security measures is proposed.

• The Journal of the Acoustical Society of Korea
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• v.33 no.1
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• pp.75-80
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• 2014

The quality of real-time Voice over Internet Protocol (VoIP) network is affected by network impariments such as delays, jitters, and packet loss. This paper proposes a packet loss concealment algorithm based on voice classification for enhancing VoIP speech quality. In the proposed method, arriving packets are classified by an adaptive thresholding approach based on the analysis of multiple features of short signal segments. The excellent classification results are used in the packet loss concealment. Additionally, linear prediction-based packet loss concealment delivers high voice quality by alleviating the metallic artifacts due to concealing consecutive packet loss or recovering lost packet.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.12
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• pp.18-26
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• 2003

In this paper, we design and implement a QoS control function in the SIP-based VoIP system. As a network infrastructure for VoIP service, we select the Intserv over Diffserv architecture where the network resources are managed by a call admission control mechanism. The SIP protocol extended to support QoS signaling procedure is modulized to operate independently with the infrastructure. The performance of the QoS-enabled VoIP system is verified by experiments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.377-384
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• 2006

According to fast VoIP technology development, more and more companies change voice network into IP based network among branch offices. IP PBX, which is deployed up to now, composed of IP phone and VoIP Gateway. Every telphone has replaced with If phone which support VoIP and VoIP gateway is installed in PBTN connection point to relay voice data. It can reduce the communication expense of International call, long distance call and call between a headquater and a trance because it uses internet line. In this paper, IP PBX is implemented that can distribute call using PBX network only usig personal terminal without Proxy Server. Depending on Role, terminal can be registered Master, Server and Client and it is verified in terms of performance and validation.

• Proceedings of the Korean Information Science Society Conference
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• 2007.10d
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• pp.619-624
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• 2007

향상된 성능의 3 세대 무선 인터넷 서비스를 제공하기 위해 등장한 IEEE 802.16 표준은 VoIP를 적용 할 수 있는 QoS 기능을 규정하고 있으며 WiMAX 포럼의 장비 제조사들은 저마다의 scheduling 기법 구현을 통해 WiMAX 네트워크에서 voice service가 가능하도록 장비를 설계하고 있다. 그럼에도 불구하고, WiMAX 무선 구간에서 사용되는 MAC header는 제한된 RF 자원의 사용에 부담을 주고 있으며, BS(base station)와 BSC(BS controller)간에 사용되는 tunneling protocol의 헤더 역시 VoIP packet에 붙게 되어 대역폭의 비효율화를 초래한다. 유선상에서의 overheader는 저렴한 Gigabit Ethernet 링크를 사용하면 대역폭이 충분히 커지므로 별 문제가 되지 않지만 무선 상의 overheader는 반드시 줄여져야만 효율적인 무선 자원 이용 및 voice quality의 향상을 가져올 수 있다. 따라서 본 연구에서는 지금까지 제안된 IP/UDP/RTP 헤더 압축 기법 및 WiMAX MAC header 압축 기법을 분석하여, 무선 구간 WiMAX 네트워크에서 VoIP 서비스를 효율적으로 제공할 수 있는 방안을 제시하고자 한다.

• Korean Journal of Human Ecology
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• v.15 no.6
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• pp.961-971
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• 2006

The purpose of this study was firstly to provide fundamental data on the activation for the IP-based video phone for the special consumer related to the physically handicapped; secondly to inform empirical data for the consumer public policy in the information technology market, specially for the deaf people. The results of study showed that consumer needs extend to not only simple voice communication for general consumers but also special demands for both the handicapped and the elderly. This study also indicated that VoIP's characteristics of technology would be easily applied to the TRS or VRS which can be adapted to the special consumer market so that VoIP service would be optimal technology for the special consumers like the deaf. In order to successfully implement TRS & VRS business, the paper proposed as follows; 1) the provision of VoIP service enable to satisfying consumers in special market such as the deaf market and the elderly market, 2) the necessity of supporting policy by the related law, and 3) the construction of the system inducing interests from the market participants.

• Journal of Internet Computing and Services
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• v.10 no.6
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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.

• Journal of KIISE:Information Networking
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• v.37 no.4
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• pp.263-271
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• 2010

SIP/RTP-based VoIP services are being popular. Recently, however, VoIP anomaly traffic such as delay, interference and termination of call establishment, and degradation of voice quality has been reported. An attacker could intercept a packet, and obtain user and header information so as to generate an anomaly traffic, because most Korean VoIP applications do not use standard security protocols. In this paper, we propose three VoIP anomaly traffic generation methods for CANCEL;BYE DoS and RTP flooding, and a detection method through flow-based traffic measurement. From our experiments, we showed that 97% of anomaly traffic could be detected in real commercial VoIP networks in Korea.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.4
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• pp.670-676
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• 2008

There is a difficult plane letting a security threat to occur in Internet networks as VoIP service uses technology-based the Internet is inherent, and you protect without adjustment of the existing security solution or changes with real-time service characteristics. It is a voice to single networks The occurrence security threat that it is possible is inherent in IP networks that effort and cost to protect a data network only are complicated relatively as provide service integrated data. This paper about various response way fields to be able to prevent analysis regarding definition regarding VoIP spam and VoIP spam technology and VoIP spam.