• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.3
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• pp.275-280
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• 2010

In this paper, VoIP(Voice over Internet Protocol) transmission performance for MANET(Mobile Ad-hoc Networks) is improved and analyzed with packet aggregation scheme which is aggregating some of short length packets to one large packet and sending to networks. VoIP simulator based on NS(Network Simulator)-2 is implemented and used to measure performance of VoIP traffic transmission. In this simulation, VoIP traffics are generated with parameters of some codes such as G.711, G.729A, GSM.AMR and iBLC. MOS(Mean Opinion Score), end-to-end network delay, packet loss rate and transmission bandwidth are measured. Performance improvements of 98% for MOS, 6.4times for end-to-end network delay, 32times for packet loss rate is shown as simulation results. On the other hand, transmission bandwidth is increased about maximum 10%. Finally, VoIP implementation guide for the performance with packet aggregation is suggested.

• 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.

• Journal of Advanced Navigation Technology
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• v.17 no.4
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• pp.421-428
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• 2013

Voice over IP refers to technology that enables routing of voice conversations over the Internet or a TCP/IP network. VoIP communication costs cheaper than traditional analog phone. Phone calls can be made to anywhere / anyone: Both to VoIP numbers as well as people with normal phone numbers. VoIP protocol equipment available today follows the SIP standard. Older VoIP equipment though would follow H 323, MGCP, Megaco/H.248. A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the TCP/IP network. A SIP server is also referred to a Asterisk IP-PBX. A VoIP telephone, also known as a SIP phone or a softphone, allows the user to make phone calls to any softphone, mobile or PC by using App store. A VoIP telephone can be a simple software-based softphone. However, the SIP Server and the program is vulnerable to VoIP attacks. In this paper, eavesdropping attacks tested by using the Asterisk SIP server. Eavesdropping attacks and TLS security methods apply to VoIP system. TLS can be applied to determine whether the eavesdropping available for VoIP Environments.

• Proceedings of the Korea Contents Association Conference
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• 2004.11a
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• pp.125-128
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• 2004

Although the current VoIP service is limited within the wire network, the future network will be a united network of wire/wireless. It is required that the study on the granting plan of numbering system by not considering the simple VoIP service but considering the entire network of wire/wireless. In this paper, we proposed a desirable VoIP service activation plan in mobile network suited to the future network by considering on VoIP service and mobile VoIP service.

• Journal of Communications and Networks
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• v.14 no.5
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• pp.510-517
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• 2012

This paper analyzes the performance of voice-over-Internet protocol (VoIP) services in terms of the system throughput, the packet delay, and the signaling overhead in a mobile WiMAX system with a hybrid automatic repeat request (HARQ) mechanism. Furthermore, a queueing analytical model is developed with due consideration of adaptive modulation and coding, the signaling overhead, and the retransmissions of erroneous packets. The arrival process is modeled as the sum of the arrival rate at the initial transmission queue and the retransmission queue, respectively. The service rate is calculated by taking the HARQ retransmissions into consideration. This paper also evaluates the performance of VoIP services in a mobile WiMAX system with and without persistent allocation; persistent allocation is a technique used to reduce the signaling overhead for connections with a periodic traffic pattern and a relatively fixed payload. As shown in the simulation results, the HARQ mechanism increases the system throughput as well as the signaling overhead and the packet delay.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.12
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• pp.20-29
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• 2014

Recently, LTE is preparing to make a new leap forward LTE-A all over the world. As LTE privides high speed service, the role of mobile phones seems to change from voice to data service. According to Cisco, global mobile data traffic will increase nearly 11-fold between 2013 and 2018. Mobile video traffic will reach 75% by 2018 from 66% in 2013 in Korea. However, voice service is still the most important role of mobile phones. Thus, controllability of throughput and low BLER is indispensable for high-quality VoIP service among various type of traffic. Although the maximum AMR-WB, 23.85 Kbps is sufficient to a VoIP call, it is difficult for the LTE which can provide tens to hundreds of MB/s may not keep the certain level VoIP QoS especially in the cell-edge area. This paper proposes a new scheduling algorithm in order to improve VoIP performance after analyzing various scheduling algorithms. The proposal is the technology which applies more priority processing for VoIP than other applications in cell-edge area based on two-tier scheduling algorithm. The simulation result shows the improvement of VoIP performance in the view point of throughput and BLER.

• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.858-861
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• 2011

최근 VoIP 서비스는 IP 네트워크의 안정화를 기반으로 국내 기업 Legacy PSTN 시장을 빠르게 대체해 가고 있다. 그러나 VoIP 서비스는 기존 인터넷망에서 발생 할 수 있는 보안 취약성 뿐 아니라 인터넷 전화 트래픽의 통과 문제 및 VoIP 스팸이나 도청 같은 기존에 없었던 새로운 이슈들을 발생 시키고 있다. 특히 SIP 인증 취약점을 이용한 Fraud Call 공격은 VoIP 서비스 사용자로 하여금 원하지 않은 호 및 과금을 대량 발생 시키는 공격기법으로 최근 기업의 피해사례가 늘어 나고 있다. 본 논문은 Fraud Call의 공격 기법을 분석하고, 호 인증 측면에서의 보안적 대응방안을 기술하고자 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2015.04a
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• pp.87-89
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• 2015

최근 빠른 속도와 사용자의 증가로 VoIP 기술에 대한 관심이 높아지고 있다. 본 논문에서는 VoIP의 개요와 VoIP 환경에서 전송 품질에 영향을 주는 요소인 지터의 정의와 지터를 개선하기 위한 지터 버퍼에 대해 설명한다. 지터 버퍼는 크게 고정형과 적응형, 두 가지로 나누어지며 지터 버퍼에 대한 분석을 통해 지터 버퍼 성능 향상 방안을 제안한다.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.4
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• pp.493-498
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• 2011

In this paper, Transmission performance over MANET(Mobile Ad-hoc Networks) under DDoS Intrusions is evaluated. Intrusion counterplan requirement, which have to be used for MANET under DDoS intrusions, is suggested through this evaluation. VoIP simulator based on NS-2 network simulator is used for performance measurement. MOS, network delay, packet loss rate and call connection rate is measured with this simulation. Finally, requirement of intrusion continuing time shorter then 10 seconds is suggested for VoIP service over MANETs under DDoS intrusions.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.543-545
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• 2011

The concern which is abundant in MANET VoIP for comprising the mobility guarantee and mobile network is received without the infrastructure system between the mobile terminal node. However, because the access of system and border is easy, the issue which is big in the security problem becomes more than the wired network system with this convenience by the foreign network attacker differently. In this paper, we would like to the fundamental web network, NAT and concluding the security problem technology in which Firewall can inquire on MANET VoIP and whether it is appropriate or not which can solve this is proposed.