• Proceedings of the Korea Multimedia Society Conference
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• 2000.11a
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• pp.299-302
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• 2000

인터넷에서 음성 서비스를 제공하는 인터넷 텔레포니 또는 VoIP(Voice over IP) 기술은 대부분 ITU-T H.323을 기반으로 제공되고 있다. 그러나 H.323은 그 구조가 복잡하기 때문에 이해하는데 상당한 노력과 오랜 개발 기간이 요구된다. IETF는 이러한 문제를 극복하고 인터넷 환경에서 잘 동작할 수 있는 IP 텔레포니용 프로토콜로서 Session Initiation Protocol (SIP)을 표준화하였다. 본 논문에서는 VoIP 서비스를 위한 SIP의 구현 사항을 기술한다

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7C
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• pp.1007-1014
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• 2004

In this paper, H.235 based security mechanism for H.323 multimedia applications was implemented in embedded environment. H.235 covers authentication using HMAC-SHAI -96, authenticated Diffie-Hellman key exchange, security capability exchange, session key management for voice encryption, and encryption functions such as DES, 3DES, RC2. H.235-based mechanisms were also analyzed in terms of its security and possible attacks.

• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.163-166
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• 2007

This paper describes a IPv6 trial service provided by LG DACOM and discusses about the output of trial service. MIC has urged public organizations to introduce IPv6 technology into their network. As one of propelling policies, MIC and NIA launched some IPv6 trial project. LG DACOM, MIC's agent in doing IPv6 trial project, has selected three public organizations in order to deploy IPv6 based VoIP trial service. KMA, KISITI and MND gave out their different service requirements. In achieve this project we developed IPv6 supported voice IP phone, video IP phone, media gateway and IP-PBX. Furthermore, two KMA provincial offices adopted trial IP phone as working phone and replaced legacy PBX with IP-PBX. At the same time, public organizations introduced IPv6 technology into their local networks.

• The Transactions of the Korea Information Processing Society
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• v.7 no.11S
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• pp.3668-3674
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• 2000

Recently Technology regarding VoIP (Voice over IP) is emerging over the market of the IP network. So far nothing is unfortunately there any attempt to try any research with respect to the development of the protocol stack relating to such control of gateway as MGCP, MEGACO, SIP, SDP. The reasons come from the low level of infrastructue, the shortage of the time and technology required at the moment, and so on. In this regards, this paper is focused on developing a protocol stack made with encoder/decoder, the generator of the header file etc, based on the protocol grammars of MGCP, SDP supported by IETF. For the sake of it, first develops the syntax analyzer, encoder/decoder, header file generator for encoding/decoding as applying the method of syntax-directed to each protocol grammar.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.1
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• pp.24-51
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• 2011

When supporting both voice and TCP in a wireless multihop network, there are two conflicting goals: to protect the VoIP traffic, and to completely utilize the remaining capacity for TCP. We investigate the interaction between these two popular categories of traffic and find that conventional solution approaches, such as enhanced TCP variants, priority queues, bandwidth limitation, and traffic shaping do not always achieve the goals. TCP and VoIP traffic do not easily coexist because of TCP aggressiveness and data burstiness, and the (self-) interference nature of multihop traffic. We found that enhanced TCP variants fail to coexist with VoIP in the wireless multihop scenarios. Surprisingly, even priority schemes, including those built into the MAC such as RTS/CTS or 802.11e generally cannot protect voice, as they do not account for the interference outside communication range. We present VAGP (Voice Adaptive Gateway Pacer) - an adaptive bandwidth control algorithm at the access gateway that dynamically paces wired-to-wireless TCP data flows based on VoIP traffic status. VAGP continuously monitors the quality of VoIP flows at the gateway and controls the bandwidth used by TCP flows before entering the wireless multihop. To also maintain utilization and TCP performance, VAGP employs TCP specific mechanisms that suppress certain retransmissions across the wireless multihop. Compared to previous proposals for improving TCP over wireless multihop, we show that VAGP retains the end-to-end semantics of TCP, does not require modifications of endpoints, and works in a variety of conditions: different TCP variants, multiple flows, and internet delays, different patterns of interference, different multihop topologies, and different traffic patterns.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.6A
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• pp.558-564
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• 2007

3G Long Term Evolution, which aims for various mobile multimedia service provision by enhanced wireless interface, proposes VoIP-based voice service through a Packet Switching (PS) domain. As delay and loss-sensitive VoIP traffic flows through the PS domain, more challenging technical difficulties are expected than in Circuit Switching (CS) domain based VoIP services. Moreover, since 3G LTE, which adopts the OFDM as its physical layer, introduces Physical Resource Block (PRB) as a unit for transmission resources, new types of resource management schemes are needed. This paper proposes a PRB scheduling algorithm of MAC layer for VoIP service in 3G LTE and shows the simulation results. The proposed algorithm has two key parts; dynamic activation of VoIP priority mode to satisfy VoIP QoS requirements and adaptive adjustment of the priority mode duration in order to minimize the degradation of resource utilization.

• 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.

• Annual Conference of KIPS
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• 2006.05a
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• pp.1235-1238
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• 2006

VoIP(Voice over IP)는 IP를 이용하여 음성과 데이터를 패킷 형태로 통합하여 실시간으로 전송하는 기술이다[1]. 본 논문에서는 VoIP망에서 리눅스 기반 IP 녹취 시스템을 설계 및 구현한다. 녹취 시스템은 고객과 상담원의 통화 내용을 자동으로 녹음하여 보관함으로써 고객의 요구 사항을 명확히 파악할 수 있다. 녹취 시스템으로 모든 네트워크 환경에서 사용할 수 있으며 CTI와 연동하여 효율적이고 체계적인 녹취 시스템 구축이 가능하다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.556-559
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• 2001

VoIP(Voice over IP) Technology, turn voice services over traditional telephone network into internet, is highlighted because of easy adopting the value added services related voice In this paper, we described the user agent system architecture for internet telephony services based on SIP (Session Initiation Protocol)

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.2 no.3
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• pp.9-16
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• 2003

In this paper, we propose the burst error reduction (BER) algorithm for VoIP service in the wireless LAN network. In end point device, this BER algorithm can be achieved packet loss bounded QoS provisioning using interleaving in buffering and FEC (Forward Error Correction) through transmitting voice packet. BER algorithm can reduced the voice packet loss rate 5.5%-60% in VoIP network using wireless LAN.