• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.331-335
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• 2007

방송과 초고속인터넷 서비스를 동시에 제공할 수 있는 HFC(Hybrid Fiber Coaxial) 망은 상/하향이 비대칭 구조이며, 하향속도에 비해 상향속도가 1/10 수준이어서 상향 트래픽이 과다하게 생성될 경우 인터넷속도 지연이 발생한다. 지연에 민감한 VoIP 서비스의 품질보장 방안으로는, DOCSIS(Data Over Cable System Interface Specification) 1.1 기반의 상향 스케쥴링 기능을 이 용한 VoCM(Voice Over Cable Modem)이 있다. 그러나 별도의 VoCM을 사용해야 하며 아날로그 전화기를 사용해 IP 기반의 VoIP 단말을 사용할 수 없다는 단점이 있다. 일반 CM(Cable Modem)에 DOCSIS 1.1 Config File을 이용하여 VoIP 품질을 보장할 경우 별도의 트래픽 대역을 항상 점유해야 하는 단점이 있다. 이에, 본 논문에서는 효율적 대역폭 이용과 단말장비에 종속적이지 않은 방안을 제안하고 일반 CM을 통한 유무선 환경하에서 Dynamic QoS(Quality Of Service)를 제공할 수 있는 PCMM(Packet Cable MultiMedia) 적용 방안 및 시험결과에 대해 고찰하고자 한다.

• Proceedings of the IEEK Conference
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• 2008.06a
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• pp.131-132
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• 2008

It is necessary to develop a high-speed wireless transmission system, which is able to send medical informations to the emergency medical center during emergency patient transportation. In this research, a system which transmits patient’s vital signs and a real-time audio/video contents of the event has been designed, developed, and the suitability of the system has been verified. Test results indicate that the system is capable of transmitting vital signal data, including 17 numeric data, 12 waveforms and 113 events, reading the affected part by forwarding a $320{\times}240$ pixel image at 2fps. Also, the full-duplex voice transmission of the system at 8bit/64kbps is enough to make stable communication between emergency medical technicians and hospital professionals possible. After numerous hours of driving, the packet loss of patient vital signs is 0.013%.

• Journal of Digital Contents Society
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• v.9 no.4
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• pp.679-686
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• 2008

Wireless LAN(WLAN) is a rather mature communication technology connecting mobile terminals. IEEE 802.11 is a representative protocol among WLAN technologies. With the rising popularity of delay-sensitive real-time multimedia applications(video, voice and data) in IEEE 802.11 wireless LAN, it is important to study the MAC layer delay performance of WLANs. In this paper, performance for packet delay model that recently have been proposed schemes is analysed in wireless LAN and proved performance results via simulation.

• Communications of the Korean Mathematical Society
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• v.14 no.2
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• pp.405-423
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• 1999

We consider a statistical multiplexer model with finite buffer capacity and finite number of independent identical 3-state bursty voice sources. The burstiness of the sources is modeled by describing both two different active periods (at the rate of one packet perslot) and the passive periods during which no packets are generated. Assuming a mixture of two geometric distributions for active period and a geometric distribution for passive period and geometric distribution for passive period, we derive the recursive algorithm for the probability mass function of the buffer contents (in packets). We also obtain loss probability and the distribution of packet delay. Numerical results show that the system performance deteriorates considerably as the variance of the active period increases. Also, we see that the loss probability of 2-state Markov models is less than that of 3-state Markov models.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4E
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• pp.161-166
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• 2003

We present an adaptive wavelet packet based speech enhancement method with robust voice activity detection (VAD) in non-stationary noise environment. The proposed method can be divided into two main procedures. The first procedure is a VAD with adaptive wavelet packet transform. And the other is a speech enhancement procedure based on the proposed VAD method. The proposed VAD method shows remarkable performance even in low SNRs and non-stationary noise environment. And subjective evaluation shows that the performance of the proposed speech enhancement method with wavelet bases is better than that with Fourier basis.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.113-118
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• 2005

Softswitch is the core BcN equipment which voice and multimedia switching based on the IP Technologies. It is designed to replace the Class 5(local Exchange) and Class 4(Toll Exchange) switch based on the circuit wired and wireless switching network technologies. Softswitch gets its name because typically it is a software based solution implemented on general purpose computers/servers. While the traditional PSTN switches are rely on dedicated facilities for T and S inter-connection and are designed primarily for voice communications. Packet based Softswitch is divided the control of call and bearer, very different from Public telephone network. Sometimes Call Agent or Media Gateway Controller, a key component in the VoIP solution, is also called Softswitch. This paper will suggest the software architecture of softswitch for performance in call processing part, also suggest the session management model to cover call setup latency.

• Proceedings of the Korea Multimedia Society Conference
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• 2012.05a
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• pp.353-354
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• 2012

광대역 VoIP 네트워크 환경에서는 불안정한 네트워크 환경으로 인해 음성패킷이 불규칙적으로 수신되어 음성데이터의 재생이 원활하지 못하다. 이러한 문제점을 해결하기 위해 본 논문에서는 네트워크 상태에 따라 원활하게 음성패킷을 재생시키는 스케줄링 방식을 제안한다. 제안하는 방식은 수신단에 도착한 패킷 헤더정보를 이용해 네트워크 지터를 추정하고, 추정된 지터와 지터버퍼와 음성프레임버퍼에 존재하는 패킷수 및 음성프레임 개수, 음성클래스정보에 따라 음성프레임의 길이를 변화시켜 재생시킴으로써 수신단의 버퍼링 지연을 줄이고 출력신호의 음성왜곡을 최소화한다. 제안하는 스케줄링 방식의 성능측정을 위해 버퍼링 지연과 PESQ를 기존 음성패킷 재생 스케줄링 방식과 비교한다.

• 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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.3 no.3
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• pp.251-265
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• 2009

VoIP service is the transmission of voice data using SIP protocol on an IP-based network. The SIP protocol has many advantages, such as providing IP-based voice communication and multimedia service with low communication cost. Therefore, the SIP protocol disseminated quickly. However, SIP protocol exposes new forms of vulnerabilities to malicious attacks, such as message flooding attack. It also incurs threats from many existing vulnerabilities as occurs for IP-based protocol. In this paper, we propose a new virtual proxy to cooperate with the existing Proxy Server to provide state monitoring and detect SIP message flooding attack with IP/MAC authentication. Based on a proposed virtual proxy, the proposed system enhances SIP attack detection performance with minimal latency of SIP packet transmission.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.8
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• pp.582-585
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• 2009

VoIP technology for transmitting voice over IP network such as packet-based network has a lot of benefits by integrating services and reducing costs. The network is different from PSTN-based communications in some aspect such as transmitting not only voice but also text, image, multimedia data. In addition, portable terminals like a mobile phone, and ubiquitous communicator can easily access the internet for VoIP. Therefore, To prevent illegal users, offering certificate services is necessary, This study proposes a solution of user certification for a VoIP environment.