• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.6
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• pp.823-829
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• 2011

This paper covers the service quality of packet-based voice service that is provided over a wireless broadband (WiBro) network. Using a measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over a wireless network[2][3], a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement result, the service quality of the voice service is supposed to be quite good over WiBro networks. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metris in which mean opinion score (MOS) starts to decrease.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.2
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• pp.249-255
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• 2012

This paper covers the service quality of packet-based voice service that is provided over a high speed downlink packet access (HSDPA) system. Using the measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over a wireless network[2][3], a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement result, the service quality of the voice service is supposed to be quite good over HSDPA system. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metrics in which mean opinion score (MOS) starts to decrease.

• Proceedings of the Korea Information Processing Society Conference
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• 2010.04a
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• pp.729-732
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• 2010

통신망의 발달로 다양한 인터넷 기반 기술들이 등장함에 따라 현재는 데이터뿐만 아닌 음성에 대한 부분도 IP 네트워크를 통해 전송하려는 움직임이 발판이 되어 VoIP(Voice Over Internet Protocol)라는 기술이 등장하였다. SIP(Session Initiation Protocol) 프로토콜 기반 VoIP 서비스는 통신 절감 효과가 큰 장점과 동시에 다양한 부가서비스를 제공하여 사용자 수가 급증하고 있다. VoIP 서비스는 호(Call)를 제어하기 위해 SIP 기반으로 구성이 되며, SIP 프로토콜은 IP 망을 이용하여 다양한 음성과 멀티미디어 서비스를 제공하게 되는데 IP 프로토콜에서 발생하는 인터넷 보안 취약점을 그대로 동반하기 때문에 DoS(Denial of Service) 및 DDoS(Distribute Denial of Service)에 취약한 성향을 가지고 있다. DDoS 공격은 단시간 내에 대량의 패킷을 타깃 호스트 또는 네트워크에 전송하여 네트워크 접속 및 서비스 기능을 정상적으로 작동하지 못하게 하거나 시스템의 고장을 유도하게 된다. 인터넷 기반 생활이 일상화 되어 있는 현 시점에서 안전한 네트워크 환경을 만들기 위해 DDoS 공격에 대한 대응 방안이 시급한 시점이다. DDoS 공격에 대한 탐지는 매우 어렵기 때문에 근본적인 대책 마련에 대한 연구가 필요하며, 정상적인 트래픽 및 악의적인 트래픽에 대한 탐지 시스템 개발이 절실히 요구되는 사항이다. 본 논문에서는 SIP 프로토콜 및 공격기법에 대해 조사하고, DoS와 DDoS 공격에 대한 특성 및 종류에 대해 조사하였으며, SIP를 이용한 VoIP 서비스에서 IP 분류와 메시지 중복 검열을 통한 DDoS 공격 탐지기법을 제안한다.

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

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.237-239
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• 2004

Wireless LAN Voice over IP(VoIP) equipment needs Quality-of-Service(QoS) with priority for processing real-time traffic. This paper shows transmit function implementation of wireless LAN(WLANs) media access control(MAC) support VoIP, and it has an advantage of guarantee of QoS and is adaptable to VoIP or mobile wireless equipment. The IEEE 802.11e standard in progress has four queues according to four access categories(AC) for transmit and the MAC transmits the data based on EDCA. The value of AC is from AC0 to AC3 and AC3 has the highest priority. The transmit method implemented at this paper ensure QoS using one transmit FIFO in hardware since real-time traffic data and non real-time traffic data has the different priority. The device driver classifies real-time data and non real-time data and transmit data to hardware with information about data type. The hardware conducts shorter backoff and selects faster AIFS slot for real-time data than it for non real-time data. Therefor It make give the real-time traffic data faster channel access chance than non real-time data and enhances QoS.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.12B
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• pp.1135-1144
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• 2003

In this paper, we proposed a semi-soft handoff mechanism to provide link mobility in IEEE 802.11 wireless LAN environment. Buffers and routing tables in APs and portals are provided in order to reroute frames, which have not been received during handoff time and have been buffered in an old AP, to a new AP after handoff is performed. For the re -routing operation, the MAC routing table should be updated by exchanging information of a mobile terminal between neighbor APs. With our proposed scheme. a wireless LAN node can perform semi soft handoff while changing its attached AP and provide mobile IP and/or real time service like voice over IP. Also, we have done simulation for evaluation of the performance of the proposed scheme. We show that our semi soft handoff mechanism can be applied for real-time service with no frame loss in mobile environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.1
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• pp.92-100
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• 2016

It is informed that the Voice over LTE(VoLTE) which serves voice and message on IP networks is better in terms of economies of scale than the legacy voice service on 2G/3G circuit-switched networks because of its technological and cost efficiency. In addition, services of voice and data are running on a single LTE network and as a result VoLTE has the more economies of scope. But, there is no study about how much technology-efficiency VoLTE has compared to circuit-based voice service and how much voice charge can be reduced as VoLTE grows up. This paper analyzes empirically cost-efficiency of VoLTE against circuit-based voice service and quantifies the reduction of voice charge as 2G/3G voice traffic shifts to VoLTE. The results describe the first is that the average cost of the total voice traffic rises shortly just after the investment of LTE network for providing VoLTE but it will soon have a capacity available to reduce the charge due to VoLTE's outstanding cost efficiency on the assumption that voice traffic is fixed, and the second is that the charge can be cut to 60% of the current rate in case of all the voice traffic moves to VoLTE. The latter proves partially the validation of data-focusing pricing plan. Our results are expected to become basic data for network operators' establishing pricing strategies and for policy makers' inducing price cutting.

• The Journal of the Korea Contents Association
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• v.2 no.3
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• pp.119-126
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• 2002

The QoS(Quality of Service) guarantee mechanism is one of critical issues in the wireless network. Real-time applications like VoIP(Voice over IP) in All-IP networks need smooth handoffs in order to minimize or eliminate packet loss as a Mobile Host(MH) transitions between network links. In this paper, we design a new multicasting algorithm using DB(Dynamic Buffering) mechanism for Mobile-IPv6. A key feature of the new protocol is the concepts of the DB and MRA(Multicast Routing Agent) to reduce delivery path length of the multicast datagram. Particularly, the number of tunneling and average routing length of datagram are reduced relatively, the multicast traffic load is also decreased.

• Management & Information Systems Review
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• v.28 no.3
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• pp.191-210
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• 2009

Internet was first designed only for sending data, but as the time passed, internet started to evolve into a broadband multi-media web that is capable of transmitting sound, video, high-capacity data and more due to the demands of internet users and the rapid changing internet-communication technology. Domestically, in January, 2000 Saerom C&T, launched a free VoIP, but due to limited ways of conversation(PC to PC) and absence of a revenue model, and bad speech quality, it had hit it's growth limit. This research studied VoIP based on technological enhancement in super-speed internet. According to IDC, domestic internet market's size was 80,800 million in 2008, and it formed a percentage of 12.5% out of the whole sound-communication market. in case of VoIP, it is able to maximize it's profit by connecting cable and wireless network, also it has a chance of becoming firm-concentrated monopoly market by fusing with IPTV. Considering the fact that our country is insignificant in MVNO revitalization, regulating organizations will play a significant roll on regulating profit between large and small businesses. Further research should be done to give VoIP a secure footing to prosper and become popularized.

• 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.