• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.261-264
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• 2011

Corded telephone to the phone using a wireless phone as the trend to switch, free Wi-Fi-enabled mobile phones, netbooks, and mobile devices, are spreading rapidly. But wireless Internet phone calls using your existing Internet network to deliver Internet services because it has a vulnerability that will occur. Government agencies are using Voice over Internet Protocol(VoIP) calls from the current wired and wireless connection and usage is increasing. In this paper, we have discovered that the vulnerability of wireless internet Wi-Fi AP and the FMC administrative agencies, such as VoIP on your wireless device to study the vulnerability. Wi-Fi AP and the FMC is to analyze the vulnerability. VoIP call interception, attack, attack on the base of the experiment is the analysis. Security-enhanced VoIP call for a Wi-Fi AP and the FMC's defense against man-in-the-middle attacks and is the study of security measures.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.1
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• pp.32-39
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• 2015

The voice communications with wireless backbone based networks are evolving into a packet switching VoIP systems. In those networks, a call processing scheme is required for management of subscribers and connection between them. A VoIP service scheme for those systems requires reliable subscriber management and connection establishment schemes, but the conventional call processing schemes based on the centralized server has lack of reliability. Thus, the mutual-backup architecture of SIP-servers is required to ensure efficient subscriber management and reliable VoIP call processing capability, and the synchronization and call processing schemes should be changed as the architecture is changed. In this paper, a mutual-backup architecture of SIP-servers is proposed for wireless backbone based networks. A message format for synchronization and information exchange between SIP servers is also proposed in the paper. This paper also proposes a FSM scheme for the fast call processing in unreliable networks to detect multiple servers at a time. The performance analysis results show that the mutual backup server architecture increases the call processing success rates than conventional centralized server architecture. Also, the FSM scheme provides the smaller call processing times than conventional SIP, and the time is not increased although the number of SIP servers in the networks is increased.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.276-285
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• 2003

In this paper, we present real-time implementation of a R2MFC/DTMF (R2 Multi Frequency Combinations/Dual Tone Multiple Frequency) tone receiver/sender using a high performance DSP (Digital Signal Processor) and apply it to a carrier class VoIP (Voice over Internet Protocol) gateway system. The Receiver utilizes the Goertzel filter and the sender adopts the harmonic resonant filter. We describe, in detail, the techniques of multi-channel real-time implementation on a Texas Instruments TMS320C62x DSP such as effective PCM (Pulse Code Modulation) in/out by means of DMA (Direct Memory Access) and McBSP (Multi Channel Buffered Serial Port) and message communication via HPI (Host Port Interface), etc. From experimental results, we confirmed that the optimized code provided 780 channel capacity at 250㎒ C6202, and the our R2MFC/DTMF receiver/sender met ITU-T (International Telecommunication Union-Telecommunication) specifications.

• Journal of KIISE:Information Networking
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• v.34 no.1
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• pp.48-61
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• 2007

WiBro (Wireless Broadband) service, developed in Korea, can provide the host mobility while its users hang around within the subnet. Next-generation Internet protocols, IPv6 and Mobile IPv6 (MIPv6), provide a plenty of addresses to the nodes and enable the handover between different subnets. However, MIPv6 is not enough to support a real time service such as VoIP (Voice over IP) due to the long latency, and it is necessary to develop an enhanced handover mechanism which is optimized to the WiBro networks. In this paper, we suggest an improved fast handover mechanism while the mobile node moves around WiBro networks. The proposal is based on Fast Mobile IPv6 (FMIPv6) which is the representative protocol for fast handover, and reduces the handover latency by the close interaction between the link layer (WiBro MAC) and IP layer (FMIPv6). Finally, we analyze the performance of proposed mechanism through the mathematical analysis.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.86-88
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• 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 Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.928-938
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• 2008

Though services requiring Quality-of-Service (QoS) guarantees such as Voice over Internet Protocol (VoIP) have been widely deployed on the internet, most of internet backbone networks, unfortunately, do not distinguish them from the best-effort services. Thus estimating the effective capacity meaning the traffic volume that the backbone networks maximally accommodate with keeping QoS guarantees for the services is very important for Internet Service Providers. This paper proposes a test-bed based on ns-2 to evaluate the effective capacity of backbone networks and then estimates the effective capacity of an experimental backbone network using the test-bed in terms of the service standard of the VoIP service. The result showed that the effective capacity of the network is estimated as between 12% and 55% of its physical capacity, which is depending on the maximum delay guarantee probability, and strongly affected by not only the type of offered workload but also the quality standard. Especially, it demonstrated that in order to improve the effective capacity the maximum end-to-end delay requirement of the VoIP service needs to be loosened in terms of the probability to guarantee.

• ETRI Journal
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• v.32 no.6
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• pp.854-862
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• 2010

One-way delay variation (OWDV) has become increasingly of interest to researchers as a way to evaluate network state and service quality, especially for real-time and streaming services such as voice-over-Internet-protocol (VoIP) and video. Many schemes for OWDV measurement require clock synchronization through the global-positioning system (GPS) or network time protocol. In clock-synchronized approaches, the accuracy of OWDV measurement depends on the accuracy of the clock synchronization. GPS provides highly accurate clock synchronization. However, the deployment of GPS on legacy network equipment might be slow and costly. This paper proposes a method for measuring OWDV that dispenses with clock synchronization. The clock synchronization problem is mainly caused by clock skew. The proposed approach is based on the measurement of inter-packet delay and accumulated OWDV. This paper shows the performance of the proposed scheme via simulations and through experiments in a VoIP network. The presented simulation and measurement results indicate that clock skew can be efficiently measured and removed and that OWDV can be measured without requiring clock synchronization.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1587-1590
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• 2002

차세대 네트워크 및 서비스를 위한 프로토콜로 IETF (Internet Engineering Task Force)의 SIP (Session Initiation Protocol)가 각광을 받고 있다. SIP는 PC, PDA, IP Phone과 같은 VoIP (Voice over IP) 단말간에 호 설정 프로토콜로 사용된다. SIP는 기본적으로는 양 단말간 호설정 프로토콜이지만 응용, 인터넷 단말기, 네트워크 장치에 구성요소로 구성할 수 있어 쉽게 적용 가능하기 때문에 모든 응용의 호설정 프로토콜로서 넓게 채택되어지고 있다. 그러나 SIP는 텍스트 기반 프로토콜로서 구현은 쉬우나 실제 표준에 맞게 구현하였는지는 판단하기가 어렵다. 따라서 구현된 SIP 프로토콜이 표준에 맞게 구현하였는지를 시험할 필요가 있다. 이를 해결하기 위해서, 본 논문에서는 SIP 시뮬레이터를 구현하였다. SIP 시뮬레이터는 구현된 SIP 제품을 인터넷상에서 시험할 수 있을 뿐만 아니라 시험 시나리오를 선택할 수 있고, 시험 과정을 그래픽하게 볼 수 있으며, 시험 결과를 확인할 수 있다. SIP 시뮬레이터는 사용자 인터페이스인 Testing User Agent와, 테스트 시나리오를 수행하는 Test Server로 구성된다. 사용자 인터페이스는 모든 플랫폼에 적용 가능한 Java를 사용하였으며, Test Server는 Linux 환경하에서 C++을 사용하여 구현하였다.

• 한국경영정보학회:학술대회논문집
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• 2007.06a
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• pp.652-657
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• 2007

인터넷의 발전으로 많은 시스템과 서비스가 변화를 거듭하고 있는 가운데, 당초 큰 기대를 모았던 인터넷전화는 저렴한 이용료 이외의 많은 장점에도 불구하고 활성화가 더디게 진행되어 왔다. 주요 원인으로 기존 통신사업자들의 입장 및 정책적인 규제 등이 지적되면서 다양한 측면에서의 활성화 방안이 논의되고 있다. 한편 기존의 통신사업자 이외에 인터넷포탈 및 방송사업자들이 시장에 새롭게 진입하면서 인터넷전화 시장은 새로운 변화를 맞이하고 있으며, 그로 인해 인터넷전화 시장은 보다 복잡한 경쟁관계를 보이고 있다. 본 연구에서는 인터넷전화 시장의 발전과정에 대한 분석을 통해 새롭게 변화하고 있는 인터넷전화 시장의 경쟁 속에서 인터넷전화 사업자들에게 전략적인 제언을 하고자 한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1602-1608
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• 2013

PoC (Push-to-Talk over Cellular) provides one-to-one as well as one-to-many communications with VoIP technology based on SIP over cellular networks. According to above property, PoC is considered as perscrptive technology for public protection for disaster relief networks. For this networks, group communication is the essential function. However, since current standardization process takes into general scenarios account without any consideration for mentioned networks, it have some problems in the point of adaptability. To solve above problem, in this paper, we propose how to reduce the overhead on the PoC server to reduce the transmission delay. Simulation results are shown to evaluate the improved performance.