• 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.

• ETRI Journal
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• v.32 no.6
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• pp.843-853
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• 2010

A successful deployment of multimedia applications over wireless environments entails improving the quality of service (QoS), not only from a technical point of view, but also considering the quality of experience (QoE) from the final user's perception. Although objective QoE measure models avoid the difficulties of subjective surveys, subjective QoE assessments are essential to understand the way users evaluate the QoS. In this work, we study the effect of a wide range of parameters on the QoE of VVoIP applications in a real wireless scenario. Through a complete statistical analysis of users' ratings, we identify the following facts. Although the use of VVoIP in wireless networks does not yet represent an advantage for users, there are great expectations for all applications under study, and with greater popularity comes higher expectations. It is easier for respondents to identify good behavior than poor behavior. Whereas the respondents' frequency of Internet use does not impact on the scores, respondents' gender does. Finally, the most determining parameters of quality from a user's perspective were instability, video quality, voice distortion, usefulness, and graphical interface.

• Annual Conference of KIPS
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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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7C
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• pp.691-699
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• 2006

Despite of the in-depth effort to cantrol the variability in IP networks, quality of service (QoS) is still not guaranteed in the IP networks. Thus, it is necessary to deal with the audible artifacts caused by packet lasses. To overcame the packet loss problem, most speech coding standard have their own embedded packet loss concealment (PLC) algorithms which adapt extrapolation methods utilizing the dependency on adjacent frames. Since many low bit rate CELP coders use predictive schemes for increasing coding efficiency, however, error propagation occurs even if single packet is lost. In this paper, we propose an efficient PLC algorithm with consideration about the speech characteristics of lost frames. To design an efficient PLC algorithm, we perform several experiments on investigating the error propagation effect of lost frames of a predictive coder. And then, we summarize the impact of packet loss to the speech characteristics and analyze the importance of the encoded parameters depending on each speech classes. From the result of the experiments, we propose a new PLC algorithm that mainly focuses on reducing the error propagation time. Experimental results show that the performance is much higher than conventional extrapolation methods over various frame erasure rate (FER) conditions. Especially the difference is remarkable in high FER condition.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.2
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• pp.57-64
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• 2011

The NGN (Next Generation Network) means the kernel infrastructure technology to provide information and communication services which are able to be used at present and future when a ubiquitous computing era has been realized. In other words, NGN can be the frame providing the same information and communication services anytime and anywhere regardless of wire and wireless. The broadband convergence network that has been built in the public institution has established a broadband multimedia communication network supporting voice telephone, task net, internet network, video conference network, voice over IP (VoIP) network and etc. It is possible for a requested bandwidth and services to be served, only if a broadband convergence network provide the interoperability between the various classes which include a transport network layer, network control layer, service control layer and other layers. In this paper, we analyzed the interoperability issues of the present broadband convergence network and propose a guideline for the future one.

• The Transactions of the Korea Information Processing Society
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• v.7 no.8S
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• pp.2722-2728
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• 2000

현재의 인터넷 망을 이용하여 음성, 화상 정보를 실시간으로 이용하고자 하는 다양한 응용이 시도되고 있다. 차세대 통신으로 주목 받고 있는 IMT-2000에서도 기존의 회선 교환망 대신 인터넷 망을 이용함으로써 경제성, 관리의 편의성, 새로운 서비스의 창출이 가능한 등의 이점이 있다. 인터넷 망이 최선의 노력(best effort)만을 제공하기 때문에 발생되는 신뢰성과 지연의 문제는 이미 많은 연구가 있어왔고 현재 어느 정도의 서비스 품질을 획득하여 VoIP(Voice Over Internet Protocol)와 같은 서비스가 실제로 이용되고 있다. 그러나 무선 통신의 경우는 이에 더하여 무선 구간에서의 자원 할당의 문제가 남아 있다. 본 연구에서는 코어 망으로 인터넷 프로토콜을 사용하는 차세대 All-IP 망에서, 무선 이동단말 간의 멀티미디어 서비스가 가능하도록 효율적인 주파수 할당을 지원하는 QoS 관리자를 설계하였다. 제안한 QoS(Quality Of Service)관리자는 요구 대역폭이 다른 멀티미디어 호 요청에 대해 융통성 있는 주파수 할당이 가능하도록 대국의 QoS 관리자와의 협상을 통해 제한된 범위 내에서 서비스 품질을 조절하여 보다 많은 호 연결 요청이 성공할 수 있도록 한다.

• Journal of KIISE:Computing Practices and Letters
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• v.11 no.4
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• pp.312-333
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• 2005

SIP(Session Initiation Protocol) has been proposed for session control protocol of Internet multimedia communication system like VoIP(Voice over IP). SIP has complicated session control steps to support various kinds of audio and video formats and to assure service quality of real time data communication. Up until now, existing protocol analyzers can not provide such detailed information of SIP based communication system. In this paper, therefore, we propose a new protocol analyzer as a tool that can analyze and diagnose SIP based multimedia communication system throughout the session initiation, data exchange and session change steps. The propose traffic analyzer, which is called STAT(SIP based Traffic Analysis Tool), Is implemented on Winder's environment so that it is generally usable and extensible. Since STAT analyze low level packets captured via Ethernet broadcasting property, it is able to provide session status and real time traffic monitoring information without any affection to the communication system. The STAT which is implemented in this paper. therefore, is expected to be a useful tool for developing and managing of a SIP based multimedia communication system.

• 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.

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