• Title/Summary/Keyword: 음성 전송 지연

Search Result 154, Processing Time 0.027 seconds

End-to-End Performance of Packet Aggregation Transmission on MANET under DDoS Attacks (DDoS 공격이 있는 MANET에서 패킷취합전송의 종단간 성능)

  • Kim, Young-Dong
    • The Journal of the Korea institute of electronic communication sciences
    • /
    • v.9 no.6
    • /
    • pp.689-694
    • /
    • 2014
  • DDoS attacks on MANET makes disable any node which support network function, cause critical results as a stopping of entire network service or separation to some parts. Packet aggregation, which gather some pieces of short length into a certain length of data, improves transmission performances on MANETs. In this paper, some effects on transmission performance of packet aggregation transmission be caused by DDoS Attacks are measured and analyzed in point of end-to-end level. MANET simulator, based on NS-2, is used for measurement end-to-end performances. MOS, connection rate, delay and packet loss rate is used as performance parameters. VoIP traffic is used for object service measurement. Finally, it is suggested that number of packet aggregation is more then 4 for keeping the transmission quality over MANETs under DDoS attacks.

End-to-End Digital Secure Speech Communication over UHF and PSTN (UHF와 PSTN간 단대단 디지털 음성보안통신)

  • Kim, Ki-Hong
    • Journal of the Korea Academia-Industrial cooperation Society
    • /
    • v.13 no.5
    • /
    • pp.2313-2318
    • /
    • 2012
  • With the widely applications of tactical radio networks, end-to-end secure speech communication in the heterogeneous network has become a very significant security issue. High-grade end-to-end speech security can be achieved using encryption algorithms at user ends. However, the use of encryption techniques results in a problem that encrypted speech data cannot be directly transmitted over heterogeneous tactical networks. That is, the decryption and re-encryption process must be fulfilled at the gateway between two different networks. In this paper, in order to solve this problem and to achieve optimal end-to-end speech security for heterogeneous tactical environments, we propose a novel mechanism for end-to-end secure speech transmission over ultra high frequency (UHF) and public switched telephone network (PSTN) and evaluate against the performance of conventional mechanism. Our proposed mechanism has advantages of no decryption and re-encryption at the gateway, no processing delay at the gateway, and good inter-operability over UHF and PSTN.

Study on Dynamic Priority Collision Resolution Algorithm in HFC-CATV Network (HFC-CATV 망에서 동적 우선순위 충돌해결알고리즘에 관한 연구)

  • Lee, Su-Youn;Chung, Jin-Wook
    • The KIPS Transactions:PartC
    • /
    • v.10C no.5
    • /
    • pp.611-616
    • /
    • 2003
  • Recently, the HFC-CATV network stand in a substructure of superhighway information network. Because of sharing up to 500 of subscribes, the Collision Resolution Algorithm needs in the upstream channel of HFC-CATV network. In order to provide Quality of Service (QoS) to users with real-time data such as voice, video and interactive service, the research of Collision Resolution Algorithm must include an effective priority scheme. In IEEE 802.14, the Collision Resolution Algorithm has high request delay because of static PNA(Priority New Access) slots structure and different priority traffics with the same probability. In order to resolve this problem, this paper proposed dynamic priority collision resolution algorithm with ternary tree algorithm. It has low request delay according to an increase of traffic load because high priority traffic first resolve and new traffic content with different probability. In the result of the simulation, it demonstrated that the proposed algorithm needs lower request delay than that of ternary tree algorithm with static PNA slots structure.

Streaming Service Scheduling Scheme in Mobile Networks (모바일환경에서 실시간 데이타서비스를 위한 스케줄링 정책)

  • Min Seung-Hyun;Kim Myung-Jun;Bang Kee-Chun
    • Journal of Digital Contents Society
    • /
    • v.3 no.1
    • /
    • pp.47-57
    • /
    • 2002
  • Recently, wireless networks have been pursuing multimedia data service as voice, data, image, video and various form of data according to development of information communication technology. It guarantees cell delivery delay of real time data in efficient real time multimedia data transfer. Also, it minimizes cell loss rate of non-real time multimedia data. In the wireless ATM, there are based on Asynchronous Transfer Mode(ATM). It implies that there are various service with difficult transmission rates and qualities in the wireless communication network. As a result, it is important to find out the ways to guarantee the Quality of Service(QoS) for each kind of traffic in wireless network. In this thesis, we propose an improved TCRM scheduling algorithms for transmission real-time multimedia data service in wireless ATM Networks. We appear real time multimedia scheduling policy that apply each different method to uplink and downlik to wireless ATM network. It can guarantee QoS requirements for each real time data and non-real time data. It also deals the fairness problem for sharing the scarce wireless resources. We solve fault of TCRM as inefficient problem of non-real data by using arbitrary transmission speed and RB(Reservation Buffer) through VC(Virtual Control) and BS(Base Station).

  • PDF

Suitable IP Currency Quality Measurement Model in Ubiquitous Environment (유비쿼터스 환경에 적합한 IP 통화품질 측정 모델)

  • Choi Seung-Kwon;Lee Byeong-Rok;Sin Byung-Gok;Kim Sun-Chul;Cho Young-Hwan
    • The Journal of the Korea Contents Association
    • /
    • v.6 no.8
    • /
    • pp.20-29
    • /
    • 2006
  • This paper proposes a quality measurement model for video phone service over IP environment. Proposed model enhances conventional E-Model by using quality analysis and this model is suitable for ubiquitous environment. This research measures video phone quality by applying bust packet loss and recency effect. It uses delay and recency effect for compensating actual quality and recognized quality of user using NR and UR factor. Simulation results show that this model can provide more precise results than conventional model by considering recency effect of video phone service quality measurement model.

  • PDF

Embedded Waveform Coding of Speech (음성 파형의 Embedded 부호화에 관한 연구)

  • 이형호;은종관
    • Journal of the Korean Institute of Telematics and Electronics
    • /
    • v.21 no.3
    • /
    • pp.73-83
    • /
    • 1984
  • The performances of embedded adaptive differential pulse code modulation (ADPCM), embedded adaptive delta modulation (ADM), and the same systems with a delayedfecision scheme have been studied with real speech over a wide dynamic range. The embedded ADPCM and ADM coders have been obtained by modifying the conventional ADPCM and ADM coders. The basic scheme of the embedded ADPCM coder is based on the ADPCM originally proposed by Cummiskey et at. For embedded ADM systems, we have modified continuously variable slope DM (CVSD) and hybrid commanding DM (HCDM) systems. Among these embedded coders, the performance of the embedded HCDM is superior to the other coders over a wide range of transmission rate from 16 to 64 kbits/s, When the delayedtecision scheme is applied to the embedded ADPCM the performance is improved significantly at all transmission rates. But, in the embedded ADM systems with 16 kHz sampling rate, the performance improvement resulting from delayed decision is not drastic as is in the embedded ADPCM with the same number of delayed samples.

  • PDF

Efficient Harmonic-CELP Based Low Bit Rate Speech Coder (효율적인 하모닉-CELP 구조를 갖는 저 전송률 음성 부호화기)

  • 최용수;김경민;윤대희
    • The Journal of the Acoustical Society of Korea
    • /
    • v.20 no.5
    • /
    • pp.35-47
    • /
    • 2001
  • This paper describes an efficient harmonic-CELP speech coder by taking advantages of harmonic and CELP coders into account. According to frame voicing decision, the proposed harmonic-CELP coder adopts the RP-VSELP coder as a fast CELP in case of an unvoiced frame, or an improved harmonic coder in case of a voiced frame. The proposed coder has main features as follows: simple pitch detection, fast harmonic estimation, variable dimension harmonic vector quantization, perceptual weighting reflecting frequency resolution, fast harmonic synthesis, naturalness control using band voicing, and multi-mode. These features make the proposed coder require very low complexity, compared with HVXC coder To demonstrate the performance of the proposed coder, a 2.4 kbps coder has been implemented and compared with reference coders. From results of informal listening tests, the proposed coder showed good quality while requiring low delay and complexity.

  • PDF

The Design and Performance Analysis of Fiber Optic Metropolitan Area Network (Fiber Optic Metropolitan Area Network의 설계 및 성능 분석)

  • 김희수;송주석
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.16 no.12
    • /
    • pp.1348-1356
    • /
    • 1991
  • The need for MAN(Metropolitan Area Network)has been increased by demands for high performance comuter communication. According to the definition of MAN by IEEE 802.6 MANs have diameter of about 50km, bandwidth of more than 1Mbps, and limited delay, Because optical fibers have unique characteristics that make them attractive for the implementation of MANs several fiber optic networks suitable for MAN application have been proposed. Those networks have drawbacks such as unlimited delay, many processing nodes and limited number of stations. Also IEEE 802.6 proposals and oters improved networks were proposed, but they have complicated access procedures and data buffering and difficulties in implementation. This paper descrives the design of Fiber Optic Metropolitan Area Network in Seoul(Seoul FOMAN) to overcome the drawback, Seoul FOMAN is hierarchical MAN and designed based on the topology of 43 end offices in Seoul. We propose MAN topology, proper access protocol and analyze the performance.

  • PDF

A Design of Multimedia Streaming Transmission Model for Continuity Guarantee based on IP (IP 기반 연속성 보장을 위한 멀티미디어 스트리밍 전송 모델 설계)

  • Kim, Hyoung-Jin;Ryu, In-Ho
    • Journal of the Korea Academia-Industrial cooperation Society
    • /
    • v.12 no.5
    • /
    • pp.2305-2310
    • /
    • 2011
  • Recently, communication industry based on data and voice and broadcasting industry centering around images have been rapidly blended. Thereupon, this article aims to suggest a multi-approach method which minimizes the use of network bandwidth allowing multimedia streaming transmission to secure IP-based continuity and let users get multimedia services of one channel or several simultaneously. Also, this study intends to design a buffering strategy that can absorb network delay and an object model to assign and maintain stable channel bandwidth.

Resource Reservation and Allocation Method for Mobile Multimedia Service (이동 멀티미디어 서비스를 위한 자원 예약 및 할당 방안)

  • 이종찬;이문호
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.29 no.7A
    • /
    • pp.766-774
    • /
    • 2004
  • The future mobile communication system can support not only voice but also multimedia applications such as data, image and video. It requires greater resources than the voice-oriented mobile system. Efficient resource reservation and hand-over schemes are necessary to maintain the same QoS of transmitted multimedia traffic because the QoS may be defected by some delay and information loss during hand-over. This paper proposes a resource reservation scheme to accommodate multimedia traffics in mobile multimedia networks. In our scheme the position of mobile is estimated in two steps, that is, sector estimation and zone estimation. With this position information, the moving direction is determined. According to simulation results, our scheme provides a better performance than conventional methods.