• Title/Summary/Keyword: VoIP phone

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Service Robot Control System using Mobile Phone (핸드폰을 이용한 서비스 로봇 제어 시스템)

  • Ahn, Ho-Seok;Sa, In-Kyu;Baek, Young-Min;Ahn, Youn-Seok;Choi, Jin-Young
    • Proceedings of the KIEE Conference
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    • 2007.10a
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    • pp.341-342
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    • 2007
  • 인터넷 전화가 보편화되면서 인터넷 전화를 이용한 많은 응용 기술이 발전하고 있다. 본 논문에서는 인터넷 전화를 이용하여 로봇을 제어하는 시스템을 소개하고자 한다. VoIP(Voice over Internet Protocol)를 이용하여 가정의 모든 가전기기를 제어한다. 특히 로봇이 전화를 받을 수 있으며, 기존의 텍스트 기반 제어 방식을 음성 기반으로 제어함으로써 시스템의 사전 학습이 없이도 쉽게 사용이 가능하다는 장점이 있다. 이 시스템은 두 종류의 로봇에 적용되어 실험하였으며, 대회에 출전하여 수상을 함으로써 객관적인 평가를 마쳤다.

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Future Trends of Blockchain and Crypto Currency: Challenges, Opportunities, and Solutions

  • Sung, Yunsick;Park, James J.(Jong Hyuk)
    • Journal of Information Processing Systems
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    • v.15 no.3
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    • pp.457-463
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    • 2019
  • The blockchain and crypto currency has become one of the most essential components of a communication network in the recent years. Through communication networking, we browse the internet, make VoIP phone calls, have video conferences and check e-mails via computers. A lot of researches are being conducting to address the blockchain and crypto currency challenges in communication networking and provide corresponding solutions. In this paper, a diverse kind of novel research works in terms of mechanisms, techniques, architectures, and frameworks have been proposed to provide possible solutions against the existing challenges in the communication networking. Such novel research works involve thermal load capacity techniques, intelligent sensing mechanism, secure cloud computing system communication algorithm for wearable healthcare systems, sentiment analysis, optimized resources.

Techniques study of IMS/SIP based Lawful Interception in 3G networks (3G 네트워크에서의 IMS/SIP 기반 합법적 감청 기법)

  • Lee, Myoung-rak;Pyo, Sang-Ho;In, Hoh Peter
    • Journal of the Korea Institute of Information Security & Cryptology
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    • v.25 no.6
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    • pp.1411-1420
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    • 2015
  • Lawful interception(LI) standard of telephone networks has technical limitations to lawfully intercept IMS/SIP-based mobile communication network subscriber who using Android and iPhone device. In addition, the technical standards related to legal interception of the IMS/SIP of the wireless network is insufficient compared to the systematic study of the development of a wireless network infrastructure. The architecture proposed in the standard of ETSI(European Telecommunications Standards Institute) for the seamless LI is insufficient to overcome the limitations of traditional voice-centric LI techniques. This paper proposes an IMS/SIP-based architecture to perform LI under 3G networks that focuses on mobility-supported environments with merging cellular networks and the Internet. We implemented the simulation to verify the efficiency of the proposed architecture, and the experimental results show that our method achieves higher lawful interception rate than that of existing interception methods.

Implementation of Variable Threshold Dual Rate ADPCM Speech CODEC Considering the Background Noise (배경잡음을 고려한 가변임계값 Dual Rate ADPCM 음성 CODEC 구현)

  • Yang, Jae-Seok;Han, Kyong-Ho
    • Proceedings of the KIEE Conference
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    • 2000.07d
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    • pp.3166-3168
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    • 2000
  • This paper proposed variable threshold dual rate ADPCM coding method which is modified from the standard ADPCM of ITU G.726 for speech quality improvement. The speech quality of variable threshold dual rate ADPCM is better than single rate ADPCM at noisy environment without increasing the complexity by using ZCR(Zero Crossing Rate). In this case, ZCR is used to divide input signal samples into two categories(noisy & speech). The samples with higher ZCR is categorized as the noisy region and the samples with lower ZCR is categorized as the speech region. Noisy region uses higher threshold value to be compressed by 16Kbps for reduced bit rates and the speech region uses lower threshold value to be compressed by 40Kbps for improved speech quality. Comparing with the conventional ADPCM, which adapts the fixed coding rate. the proposed variable threshold dual rate ADPCM coding method improves noise character without increasing the bit rate. For real time applications, ZCR calculation was considered as a simple method to obtain the background noise information for preprocess of speech analysis such as FFT and the experiment showed that the simple calculation of ZCR can be used without complexity increase. Dual rate ADPCM can decrease the amount of transferred data efficiently without increasing complexity nor reducing speech quality. Therefore result of this paper can be applied for real-time speech application such as the internet phone or VoIP.

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The Development of 1G-PON Reach Extender based on Wavelength Division Multiplexing for Reduction of Optical Core (국사 광역화와 광코어 절감을 위한 파장분할다중 기반의 1기가급 수동 광가입자망 Reach Extender 효율 극대화 기술 개발)

  • Lee, Kyu-Man;Kwon, Taek-Won
    • Journal of Digital Convergence
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    • v.17 no.8
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    • pp.229-235
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    • 2019
  • As the demand for broadband multimedia including the Internet explosively increases, the advancement of the subscriber network is becoming the biggest issue in the telecommunication industry due to the surge of data traffic caused by the emergence of new services such as smart phone, IPTV, VoIP, VOD and cloud services. In this paper, we have developed WDM(Wavelength Division Multiplexing)-PON(passive optical network) based on the 1-Gigabit Reach Externder (RE) technique to reduce optical core. Particularly, in order to strengthen the market competitiveness, we considered low cost, miniaturization, integration technique, and low power of optical parts. In addition, we have developed a batch system by integrating all techniques for reliability, remote management through the development of transmission distance extension and development of capacity increase of optical line by using RE technology in existing PON network. Based on system interworking with existing commercial 1G PON devices, it can be worthy of achievement of wide nationalization and optical core reduction by using this developed system. Based on these results, we are studying development of 10G PON technology.

Distributed processing for the Load Minimization of an SIP Proxy Server (SIP 프록시 서버의 부하 최소화를 위한 분산 처리)

  • Lee, Young-Min;Roh, Young-Sup;Cho, Yong-Karp;Oh, Sam-Kweon;Hwang, Hee-Yeung
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.9 no.4
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    • pp.929-935
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    • 2008
  • As internet telephony services based on Session initiation Protocol (SIP) enter the spotlight as marketable technology, many products based on SIPs have been developed and utilized for home and office telephony services. The call connection of an internet phone is classified into specific call connections and group call connections. Group call connections have a forking function which delivers the message to all of the group members. This function requires excessive message control for a call connection and creates heavy traffic in the network. In the internet cail system model. most of the call-setup messages are directed to the proxy server during a short time period. This heavy message load brings an unwanted delay in message processing and. as a result, call setup can not be made. To solve the delay problem, we simplified the analysis of the call-setup message in the proxy server, and processed the forking function distributed for the group call-setup message. In this thesis, a new system model to minimize the load is proposed and the subsequent implementation of this model demonstrates the performance improvement.

The Performance Improvement of G.729 PLC in Situation of Consecutive Frame Loss (연속적인 프레임 손실 상황에서의 G.729 PLC 성능개선)

  • Hong, Seong-Hoon;Kim, Jin-Woo;Bae, Myung-Jin
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.1
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    • pp.34-40
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    • 2010
  • As internet spread widely, various service which use the internet have been provided. One of the service is a internet phone. Its usage is increasing by the advantage of cost. But it has a falling off in quality of speech. because it use packet switching method while existing telephone use circuit switching method. Although vocoder use PLC (Packet Loss Concealment) algorithm, it has a weakness of continuous packet loss. In this paper, we propose methods to improve a lowering in quality of speech under continuous loss of packet by using PLC algorithm used in advanced G.729 and G.711. The proposed methods are LP (Linear Prediction) parameter interpolation, excitation signal reconstruction and excitation signal gain reconstruction. As a result, the proposed method shows superior performance about 11%.

Wireless Webcam Implementation and Application Utilizing Wireless USB Technology (무선 USB 기술을 활용한 무선웹캠 구현 및 적용방법)

  • Chae, Jung-Sik;Ban, Tae-Hak;Jung, Hoe-Kyung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.18 no.3
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    • pp.569-575
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    • 2014
  • These days as smart TVs or android set top boxes have come into wide use, VoIP(Voice over Internet Protocol) services such as the Skype which can beused on computer system, can be also accessed by smart TVs or set top boxes. and the various kinds of distance learning programs have been developed on smart TVs or set top boxes. but when users have utilized a webcam with smart TVs or set top boxes, users have to go in front of smart TVs or set top boxes In this research paper, the new technology can be solved the problem of webcam use according to chaned environment. This tech is related with the wireless webcam one which can be applicable to open spaces such as living room with some distance between user and smart TV. There are few problems in long distance the test results show high reliability in living room with some distance.