• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.905-908
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• 2011

와이파이 기능이 탑재된 모바일 기기 보급이 확산되면서 무선네트워크를 이용한 많은 서비스가 개발되고 있다. 그중 기존 전화망(PSTN)에서 발전하여 인터넷 네트워크를 이용한, 음성과 데이터 네트워크 융합의 대표적인 인터넷 전화(VoIP)서비스 이용률이 증가하고 있는 추세다. VoIP 기술은 FMC(Fixed Mobile Convergence) 서비스의 기반이 되며, 이에 따라 FMC서비스는 기존의 VoIP 보안위협 및 특성을 상속 받게 된다. 본 논문은 유무선 통합에 의한 여러 가지 유무선 단말, 네트워크 및 서비스 특성에 대한 보안 위협을 상속 받게 되는 FMC 환경에서의 VoIP보안 위협을 소개하고 보안 요구사항을 설계한다. 또한 안전한 FMC서비스를 위해 총체적인 보안망 설계 시 VoIP보안 위협 및 보안요구사항에 적합한 보안솔루션의 아키텍처 구조를 제안한다.

• Journal of Korea Multimedia Society
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• v.10 no.4
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• pp.450-461
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• 2007

In this paper, we propose the method to use power-saving mode (PSM) applicable to non real-time traffic and PSM applicable to real-time traffic simultaneously, for VoIP traffic with silence suppression. The proposed method uses PSC II during talk-spurt interval of parties A and/or B and uses PSC I or probabilistic sleep interval decision (PSID) method during mutual silence interval, respectively. To evaluate the performance of hybrid PSM (HPSM) based on PSC II or PSID method, we present average buffering delay, energy consumption of mobile station and VoIP packet drop probability with simulation runs. Results shows that proposed HPSM decreases energy consumption of mobile station up to 25 % while satisfying the packet drop probability within QoS requirement in case of end-to-end VoIP connection.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.3B
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• pp.304-310
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• 2009

In this paper, we evaluate the performance of VoIP in OFDMA-based relay systems with various capabilities of relays. We classify relays according to capability as "mid-capability (MC)" and "high-capability (HC)" relay. In system with HC relays, not only base station (BS) but also relay station (RS) performs scheduling at its ova whereas only BS performs scheduling in system with MC relays using the information reported by MS (i.e. the received signal-to-interference-plus-noise ratio (SINR) of mobile station (HS), the amount of MS traffic, etc). In system with MC relays, the controling overhead of BS is larger than that of system with HC relays. However, since BS has all MS information, efficient resource allocation and scheduling is possible. We derived the "average packet delay," "good packet ratio," and "cell goodput" in a VoIP environment. The simulation results demonstrate that the system with MC relays has better VoIP performance over that with HC relays.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.2
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• pp.257-262
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• 2012

In this paper, transmission performance for serval VoIP traffics is meaured and analysised over MANET(Mobile Ad-hoc Networks) under some types of intrusions. VoIP simulator based on NS-2 is used for performance measure. In the simulation, MOS, network delay, packet loss rate and call connection rate is measured for transmission performance. With analysis of measued data, characteristics of each traffics for transmission performance parameters is showed on MANET under multi intrusions, As a results of this paper, some recommendation condtion for codec selection and blackhole intrusion is suggested to implement VoIP services on MANET under multi intrusions like as DDoS and blockhole intrusions.

• Journal of information and communication convergence engineering
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• v.7 no.3
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• pp.309-313
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• 2009

In this paper, end-to-end VoIP(Voice over Internet Protocol) performance is evaluated by simulation with NS-2 simulation tool. There are many results studied and published for VoIP performance over TCP/IP networks. But, almost all of them were focused on wired or wireless Internet environments. About MANET (Mobile Ad Hoc Network), VoIP is currently studying several points of research. In this paper, analysis of VoIP performance is done with focusing on the mobility of MANETs. MOS(Mean Opinion Score), network delay, packet loss rates are considered as end-to-end QoS performance parameters.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.277-280
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• 2011

VoIP is a technology of voice communication, using the existing internet network which sends and receives voice packets. VoIP has an advantage that VoIP is cheaper than an existing telephony, and the tech is vitalized lately. But recently you can download Volp Application in the Market that have a vulnerability(Anyone Can Upload). This weakness is wrongfully used that People are downloaded by encouraging about malignant code is planted. Signal intercepts indicates from this case. and paralysis by DDoS Attack, bypass are charged for hacking. Judging from, security threat of VolP analysis and take countermeasures. In the thesis we analyze the VoIP security caused on 'Soft Phone' and 'Smart Phone', and figure out security policies and delineate those policies on the paper.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.1
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• pp.71-76
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• 2013

In this paper, end-to-end performances for some VoIP traffics are measured and analyzed over MANET(Mobile Ad-hoc Networks) with packet aggregation transmission under blackhole attacks. A VoIP simulator based on NS-2 is used for performance measure. In the simulation, MOS, call connection ratio, network delay and packet loss rate are measured for end-to-end transmission performance. With analysis of measured data, characteristics of each traffics for performance parameters of packet aggregation transmission is showed on MANET under blackhole attack, As a results of this paper, some considerations be required for packet aggregation transmission are suggested to implement VoIP services on MANET under blackhole attacks.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.415-424
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• 2008

In this paper, we compare the QoS performance enhancement schemes according to the network architecture of mobile WiMAX system in order to improve the end-to-end QoS performance and propose QoS parameter mapping method and IP packet scheduling algorithm. To evaluate the end-to-end QoS performance, we implemented an end-to-end simulator of mobile WiMAX system using OPNET. Simulation results show that the proposed QoS parameter mapping scheme reduces the average delay of VoIP packet and improves uplink resource efficiency. And also, when the proposed IP packet scheduling algorithm is applied to the system, the end-to-end packet transmission delay of VoIP service can be reduced by 44-67 percent compared to FIFO and WRR scheduler.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.398-407
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• 2010

Due to increase of user requirement for various traffics and the advance of network technology, each distinct network has converge into FMC(Fixed Mobile Convergence) networks. However, we need to research the performance analysis of VoIP(Voice over Internet Protocol) in the FMC network to provide QoE for the voice user of FMC network. Therefore, this paper introduces the scenario which is the situation of voice quality degradation when a user uses VoIP to communicate with other users in the FMC network. Especially, this paper presents scenario in terms of the component of the network and finds the improvement point of voice quality. In the simulation results, three improvement points of voice quality are found as following: voice quality degradation by packet loss in the physical layer of the HSDPA network, by utilizing GGSN without QoS parameter mapping mechanism which is gateway between 3GPP and IP backbone, and by using non-QoS AP in the WLAN network.