• Journal of Satellite, Information and Communications
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• v.8 no.4
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• pp.53-57
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• 2013

The quality of service for the voice and data communication is one of the most important elements in the mobile cellular networks. This quality has the time varying characteristics depending on the wireless network environments. In order to guarantee the quality of service at the predefined level, the mobile service provider needs to collect all the information about the level of quality served by the local base stations. In the conventional method the information on the service quality is measured and collected by the moving vehicle adopting the global positioning device. However this method requires relatively high cost and does not cover all the service areas. In this paper a new method is proposed utilizing a measurement node with local area communication transceiver and the mobile user equipment. In the proposed method the node communicates with the user equipment periodically to monitor the information on the service quality. The resulting information is reported to the measurement server in the mobile network. This method could be applied to either the circuit or the packet based networks. It also has an advantage in terms of the cost efficiency compared to the conventional method considering the required the equipment and the human resources.

• Journal of Advanced Navigation Technology
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• v.28 no.3
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• pp.309-314
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• 2024

Avionics Intercom performs communication between pilot, co-pilot and crews in aircraft. In the case of developing intercom in the aircraft modification development project, additional communication equipment or avionics equipment is configured to link the existing intercom with the headsets. Newly designed intercom needs a configuration that receives an aircraft headset microphone and transmits a microphone signal to the existing intercom, and these signals are required to perform signal quality above a certain level. To satisfy these requirements, microphone transmitter circuit has designed and tested, but quality factors of signal were not suitable. In order to avoid the issue, eliminate transmitter and apply signal bridge circuit considered with load effect, and it meets requirements. In this paper, the test results for the signal quality for each configuration are reviewed.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.4
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• pp.161-167
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• 2015

Today, computer network infrastructure is commonplace and internet user increase. Transmission bandwidth for network traffic is increasing. The data amount of 2014 and 2015 year measured 47 and 62 exabyte, it shows a tremendous increase. Video traffic occupy 60 percent among internet traffic. Enormous growth of Internet traffic than the network infrastructure is not enough to support them, so mobile network occur side-effects such as fall of voice communication. In this paper, we proposed the video compression method using SURF feature and block shaping in order to avoid potential damage. Our proposed method can provide enough performance than previous method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.12
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• pp.2555-2562
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• 2011

Binary CDMA is a new standard technology for wireless communication developed by our country that makes high speed communications and good quality of services. In this paper we researched the development methods and implementation of a wireless VoIP System based on the Binary-CDMA technology that makes it freely installed in any place without phone cables and laying works. We designed the structure of the Binary-CDMA Wireless VoIP System and defined messages and data structures for call control. And then we designed and implemented call control message flow between VoIP terminals for user's communication.

• Journal of KIISE:Information Networking
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• v.37 no.4
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• pp.263-271
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• 2010

SIP/RTP-based VoIP services are being popular. Recently, however, VoIP anomaly traffic such as delay, interference and termination of call establishment, and degradation of voice quality has been reported. An attacker could intercept a packet, and obtain user and header information so as to generate an anomaly traffic, because most Korean VoIP applications do not use standard security protocols. In this paper, we propose three VoIP anomaly traffic generation methods for CANCEL;BYE DoS and RTP flooding, and a detection method through flow-based traffic measurement. From our experiments, we showed that 97% of anomaly traffic could be detected in real commercial VoIP networks in Korea.

• Electronics and Telecommunications Trends
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• v.18 no.2 s.80
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• pp.79-88
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• 2003

무선인터넷 서비스는 이동통신 사업자들에게 강력한 추가수익원으로 떠오르고 있으며, 활성화를 위하여 장기적이고 효율적인 정책대안 수립이 절실한 시점이다. 본 연구에서는 무선인터넷 산업의 현실을 올바로 파악하고자 하는 노력의 일환으로 무선인터넷 사용자들에 대하여 설문조사를 수행하였으며, 그 결과를 무선인터넷 이용환경, 이용실태, 평가 등 세 부분으로 나누어 서술하였다. 무선인터넷은 엔터테인먼트용 콘텐츠의 이용도가 높아 여가생활에 도움을 주고 있지만, 실생활에는 아직 큰 도움이 되지 못한다는 평가를 얻었다. 그리고 통신요금에 대한 불만이 가장 높은 것으로 나타나, 무선인터넷 서비스 확산에 큰 장애요인으로 떠올랐다. 그리고 정액제에 대한 높은 선호를 확인함으로써 통신요금 체계도 무선인터넷 확산에 걸림돌이 되고 있는 것으로 판단하였다. 한편 본 조사에서는 전자우편의 이용횟수, 무선인터넷 및 음성통화의 하루평균 이용시간, 통신요금 및 정보이용료 지출액 등에서 여성의 활용도가 활발한 것으로 나타나 이동통신 서비스 확산에서 여성의 역할이 부각되었다. 결론적으로 우리나라는 무선인터넷 서비스 확산을 위한 하드웨어적인 준비는 거의 완료되었으나 요금수준 및 체계, 통신품질, 콘텐츠 등 소프트웨어 측면에서의 준비는 다소 미비한 것으로 판단되며, 해당 분야에 대한 집중적인 연구와 투자가 필요한 시점으로 생각된다.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.479-484
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• 2003

In the current Internet, it is difficult for an Internet Phone to guarantee the QoS due to variable network conditions such as packet loss rate, delay and bandwidth. In addition, the QoS of an Internet Video Phone is more hard to guarantee because of video data. In this paper, we investigate application-level QoS control schemes that can adapt to variable network conditions, and describe an error control scheme and a congestion control scheme. Based on these QoS control schemes, we have designed and implemented an Internet Video Phone System that supports adaptive audio and video delivery. Through experiments, we found that the Internet Video Phone can reduce the packet loss rate considerably as well as adjust the transmission rate considering other TCP flows.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.405-414
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• 2002

In this paper, we propose a new objective speech quality measure using noise masking threshold for speech quality assessment of mobile telephone network environments, and verify the effectiveness of the proposed method through the experiments. For such a purpose, well known objective speech quality measures such as BSD and PSQM are first evaluated for digital mobile telephone network environments. However, these conventional methods does not have good performance under mobile networks environments compared to literary results. To be mote effective objective speech quality measure under mobile telephone environments, the proposed method employs human psychoacoustic masking effect. The DMOS, instead of MOS, is used as a subjective speech quality measure for performance evaluation. The performance comparison are carried out with speech data collected from digital mobile telephone environments. As results, the proposed measure have and average 4% higher performance, in terms of correlation, than existing objective speech quality measures such as BSD and PSQM.

• 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.

• Journal of Internet Computing and Services
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• v.10 no.6
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• pp.27-36
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• 2009

PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.