• Proceedings of the KIEE Conference
• /
• 2004.11c
• /
• pp.86-88
• /
• 2004

As the rapid incensement of the number of internet users has occurred recently, many multimedia application services have been emerging. To improve quality of service, traffic can be suggested to be classified with priority in EPON(Ethernet Passive Optical Network), which is supporting the multimedia application services. In this paper, multimedia application services treat bandwidth classifying device in serving both delay sensitive traffic for real-time audio, video and voice data such as VoIP(Voice over Internet Protocol), and nonreal-time traffic such as BE(Best Effort). With looking through existing mechanisms, new mechanism to improve the quality will be suggested. The delay performances and packet losses of traffic achieved by supporting bandwidth allocation of upstream traffic in suggested mechanism will be analyzed with simulation.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.9 no.6
• /
• pp.695-701
• /
• 2014

Blackhole attacks having a unauthorized change of routing data will cause critical effects for transmission performance. The transmission performance will be improved to the a certain level by using or having IDS(Intrusion Detection System)/IPS(Intrusion Prevention System) as countermeasures to blackhole attacks. In this papar, the effects of IDS to ene-to-end performance of packet aggregation transmission are analyzed on MANET(Mobile Ad-hoc Network) with IDS under blackhole attacks. MANET simulator based on NS-2 is used to analyze performance parameters as MOS, connection ratio, delay and packet loss rate as standard performance parameters, an another performance factor which is suggested in this paper. VoIP(Voice over Internet Protocol) traffics, one of voice services, is used for performance analysis. A suggestion for IDS implementation on MANET with packet aggregations under blackhole is shown as one of results.

• Journal of Communications and Networks
• /
• v.11 no.1
• /
• pp.72-83
• /
• 2009

We have studied the effect of head of the line blocking (HOLB) on session initiation protocol (SIP) session establishment delays. Our results are based on experiments performed in a test bed and on the public Internet. We used the stream control transmission protocol (SCTP) as a transport for SIP because SCTP can be configured to suffer or to avoid HOLB. Our experiments show that the effect of HOLB on session establishment delays generally starts to be significant starting at fairly low packet loss rates. However, there are scenarios where network conditions are good enough to make the effect of HOLB insignificant.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.4A
• /
• pp.310-315
• /
• 2002

The Bark Coherence Function (BCF) defies a coherence function within perceptual domain as a new cognition module, robust to linear distortions due to the analog interface of digital mobile system. Our previous experiments have shown the superiority of BCF over current measures. In this paper, a new BCF suitable for VoIP is developed. The unproved BCF is based on the wavelet series expansion that provides good frequency resolution while keeping good time locality. The proposed Wavelet based Bark Coherence function (WBCF) is robust to variable delay often observed in packet-based telephony such as Voice over Internet Protocol (VoIP). We also show that the refinement of time synchronization after signal decomposition can improve the performance of the WBCF. The regression analysis was performed with VoIP speech data. The correlation coefficients and the standard error of estimates computed using the WBCF showed noticeable improvement over the Perceptual Speech Quality Measure (PSQM) that is recommended by ITU-T.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.8 no.2
• /
• pp.634-645
• /
• 2014

VoIP (Voice over Internet Protocol) is a technology of transmitting and receiving voice and data over the Internet network. As the telecommunication industry is moving toward All-IP environment with growth of broadband Internet, the technology is becoming more important. Although the early VoIP services failed to gain popularity because of problems such as low QoS (Quality of Service) and inability to receive calls as the phone number could not be assigned, they are currently established as the alternative service to the conventional wired telephone due to low costs and active marketing by carriers. However, VoIP is vulnerable to eavesdropping and DDoS (Distributed Denial of Service) attack due to its nature of using the Internet. To counter the VoIP security threats efficiently, it is necessary to develop the criterion or the model for estimating the information security level of VoIP service providers. In this study, we developed reasonable security indicators through questionnaire study and statistical approach. To achieve this, we made use of 50 items from VoIP security checklists and verified the suitability and validity of the assessed items through Multiple Regression Analysis (MRA) using SPSS 18.0. As a result, we drew 23 indicators and calculate the weight of each indicators using Analytic Hierarchy Process (AHP). The proposed indicators in this study will provide feasible and reliable data to the individual and enterprise VoIP users as well as the reference data for VoIP service providers to establish the information security policy.

• Proceedings of the KIEE Conference
• /
• 2002.07d
• /
• pp.2316-2318
• /
• 2002

지금까지 디지털 통신에서는 반향이 통신품질의 관점에서 별다른 문제가 되지 않았다. 그러나 인터넷의 발달로 인하여 음성 데이터 통합(VoIP:Voice over Internet Protocol)을 이용한 인터넷폰의 사용이 요구되고 있으며, 시외 또는 국제 통화의 경우에 음성신호를 서킷에서 패킷으로 전송하는 과정에서 전송 지연 증가에 따른 반향에 대한 문제가 발생되고 있다. 본 논문에서는 VoIP 기반의 음성통신에서 발생하는 반향을 적응 반향제어기를 통해 제거하는 방법에 대해 연구하였다. 모의 실험을 통해 ECLMS 알고리즘을 적용한 반향제거기가 우수한 반향제거 성능을 보여줌을 확인하였다.

• Proceedings of the KIEE Conference
• /
• 2003.07d
• /
• pp.2127-2129
• /
• 2003

지금까지 디지털 통신에서는 반향이 통신 품질의 관점에서 별다른 문제가 되지 않았다. 그러나 인터넷의 발달로 인하여 음성 데이터 통합(VoIP:Voice over Internet Protocol)을 이용한 인터넷폰의 사용이 요구되고 있으며, 시외 또는 국제 통화의 경우에 음성 신호를 서킷에서 패킷으로 전송하는 과정에서 전송 지연 증가에 따른 반향에 대한 문제가 발생되고 있다. 현재는 DSP chip의 급속한 발달로 반향의 제거가 실시간으로 처리할수 있게 되었다. 본 논문에서는 VoIP기반의 음성 통신에서 발생하는 반향을 적응 반향제어기를 통해 제거하는 방법에 대해 연구하였다. DSP processor를 사용한 실험을 통해 알고리즘을 적용한 반향제거기의 성능이 우수함을 확인하였다.

• Journal of information and communication convergence engineering
• /
• v.7 no.3
• /
• pp.309-313
• /
• 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
• /
• v.30 no.2A
• /
• pp.12-19
• /
• 2005

The VoIP(Voice over IP) has been worldwide used and already put to practical use in many fields. However, it is needed to ensure secret of VoIP call in a special situation. It is relatively difficult to eavesdrop the commonly used PSTN in that it is connected with 1:1 circuit. However, it is difficult to ensure the secret of call on Internet because many users can connect to the Internet at the same time. Therefore, this paper suggests a new model of Internet telephone for eavesdrop prevention enabling VoIP(using SIP protocol) to use the VPN protocol and establish the probability of practical use comparing it with Internet telephone.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.8B
• /
• pp.797-804
• /
• 2009

Recently, SIP(Session Initiation Protocol) is used to set up and manage sessions for multimedia applications such as VoIP(Voice over IP) and IMS(IP Multimedia Subsystem). However, because SIP operates over the Internet, it is exposed to pre-existed internet security threats such as service degradation or service disruptions. Multimedia applications which are delay sensitive even suffers more from the threats mentioned above. The proposed methods so far to detect SIP INVITE flooding are CUSUM(Cumulative Sum), Hellinger distance and adaptive threshold, but among methods only take normal state into consideration. So, it is not capable of adapting the condition of the network congestion which are dynamically changing. In this paper, SIP INVITE flooding detection algorithm considering network congestion which enables efficient detections of such attacks is proposed. The proposed algorithm is expected to detect other types of attacks such as BYE and CANCEL more precisely compared to other methods.