• Title/Summary/Keyword: Packet error rate

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Implementation of Adaptive Transmission Middleware for Video Streaming (비디오 스트리밍을 위한 적응적 전송 미들웨어의 구현)

  • 김영주
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.8 no.3
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    • pp.637-644
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    • 2004
  • This paper proposed and implemented the adaptive transmission middleware for video streaming, which is able to support the adaptive transmission of video data to the fluctuating changes of network environment in the packet-based network and the properties of transmitted video data. The adaptive transmission middleware is made up SR-RTP-based transfer module and TFRC(TCP Friendly Rate Control)-based transfer-rate control module. The SR-RTP-based transfer module supports RTP-based real-time transfer of video data and packet retransmission scheme retransmitting the high-priority packets selectively in the damaged video data to reduce the error induced by the packet loss. Sharing the transmission bandwidth of network with the TCP-based data transfer, the TFRC-based transfer-rate control module controls the transfer rate of video data according to the most allowable transmission bandwidth in the network, so that the transfer rate is controlled adaptively to the fluctuating changes of transmission bandwidth. This paper, for the experiment, applied the adaptive transmission middleware to video streaming in the external Internet environment, and analyzed the effective frame transfer rate and the degree of the streaming jitter to evaluate the performance of packet-loss recovery and adaptive transfer rate control. In the external Internet environment where the packet-loss rate is high a bit, the relatively high streaming performance was showed compared with the case that didn't apply the adaptive transmission middleware.

Endowment of Duplicated Serial Number for Window-controlled Selective-repeat ARQ (Window-controlled Selective-repeat ARQ에서 중복된 순차 번호의 부여)

  • Park, Jin-Kyung;Shin, Woo-Cheol;Ha, Jun;Choi, Cheon-Won
    • Journal of IKEEE
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    • v.7 no.2 s.13
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    • pp.288-298
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    • 2003
  • We consider a window-controlled selective-repeat ARQ scheme for error control between two adjacent nodes lying on a communication path. In this scheme, each packet to be transmitted is endowed with a serial number in a cyclic and sequential fashion. In turn, the transmitting node is not allowed to transmit a packet belonging to a window before every packet in the previous window is positively acknowledged. Such postponement of packet transmission incurs a degradation in throughput and delay performance. In this paper, aiming at improving packet delay performance, we employs a supplement scheme in which a serial number is duplicated within a frame. Classifying duplication rules into fixed, random and adaptive categories, we present candidate rules in each category and evaluate the packet delay performance induced by each duplication rule. From numerical examples, we observe that duplicating serial numbers, especially ADR-T2 effectively reduces mean packet delay for the forward channel characterized by a low packet error rate. We also reveal that such delay enhancement is achieved by a high probability of hitting local optimal window size.

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Low-Delay LSF FEC Technique Robust in Lossy VoIP Environment (VoIP 손실 환경에 강인한 저지연 LSF FEC 기법)

  • Yang, Hae-Yong;Lee, Kyung-Hoon;Hwang, In-Ho
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.39 no.6
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    • pp.687-695
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    • 2002
  • Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.

HWbF(Hit and WLC based Firewall) Design using HIT technique for the parallel-processing and WLC(Weight Least Connection) technique for load balancing (병렬처리 HIT 기법과 로드밸런싱 WLC기법이 적용된 HWbF(Hit and WLC based Firewall) 설계)

  • Lee, Byung-Kwan;Kwon, Dong-Hyeok;Jeong, Eun-Hee
    • Journal of Internet Computing and Services
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    • v.10 no.2
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    • pp.15-28
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    • 2009
  • This paper proposes HWbF(Hit and WLC based Firewall) design which consists of an PFS(Packet Filter Station) and APS(Application Proxy Station). PFS is designed to reduce bottleneck and to prevent the transmission delay of them by distributing packets with PLB(Packet Load Balancing) module, and APS is designed to manage a proxy cash server by using PCSLB(Proxy Cash Server Load Balancing) module and to detect a DoS attack with packet traffic quantity. Therefore, the proposed HWbF in this paper prevents packet transmission delay that was a drawback in an existing Firewall, diminishes bottleneck, and then increases the processing speed of the packet. Also, as HWbF reduce the 50% and 25% of the respective DoS attack error detection rate(TCP) about average value and the fixed critical value to 38% and 17%. with the proposed expression by manipulating the critical value according to the packet traffic quantity, it not only improve the detection of DoS attack traffic but also diminishes the overload of a proxy cash server.

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Comparison about TCP and Snoop protocol on wired and wireless integrated network (유무선 혼합망에서 TCP와 Snoop 프로토콜 비교에 관한 연구)

  • Kim, Chang Hee
    • Journal of Korea Society of Digital Industry and Information Management
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    • v.5 no.2
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    • pp.141-156
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    • 2009
  • As the TCP is the protocol designed for the wired network that packet loss probability is very low, because TCP transmitter takes it for granted that the packet loss by the wireless network characteristics is occurred by the network congestion and lowers the transmitter's transmission rate, the performance is degraded. The Snoop Protocol was designed for the wired network by putting the Snoop agent module on the BS(Base Station) that connect the wire network to the wireless network to complement the TCP problem. The Snoop agent cash the packets being transferred to the wireless terminal and recover the loss by resending locally for the error occurred in the wireless link. The Snoop agent blocks the unnecessary congestion control by preventing the dupack (duplicate acknowledgement)for the retransmitted packet from sending to the sender and hiding the loss in the wireless link from the sender. We evaluated the performance in the wired/wireless network and in various TCP versions using the TCP designed for the wired network and the Snoop designed for the wireless network and evaluated the performance of the wired/wireless hybrid network in the wireless link environment that the continuous packet loss occur.

Throughput Performance of Hybrid ARQ Ultra-Wideband Communication System for Wireless Packet Transmission (무선 패킷 전송을 위한 Hybrid ARQ 광대역 통신시스템의 처리율 성능)

  • Roh, Jae-Sung
    • Journal of Advanced Navigation Technology
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    • v.11 no.3
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    • pp.274-280
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    • 2007
  • An ultra-wideband signal is characterized by a radiated spectrum with wide bandwidth around a relatively low center frequency. In this paper, the bit error rate (BER), packet error rate (PER), and data throughput performance for an ultra-wideband system with M-ary correlation receiver are analyze in additive white Gaussian noise (AWGN) and co-channel interference channel. To evaluate the performance of UWB system, a set of UWB communication waveform as pulse position modulated (PPM) signals consisting of more than one UWB pulse is used. The M-ary PPM signals are defined to be equally correlated in order to simplify the system performance analysis. The analysis for system performance shows that the wireless channel error significantly degrades throughput performance and can be effectively increased by hybrid ARQ scheme. Also, an attempt for comparing the data throughput of ultra-wideband system on different performance improvement schemes and parameters has been made. From the performance evaluation process, it is shown that the effects of wireless channel and hybrid ARQ scheme for ultra wideband M-ary PPM system can be evaluated by means of a suitable combination of the PER, throughput vs. signal-to-noise power ratio per bit.

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Design of MAC Protocol to Guarantee QoS for Multimedia Traffic in a Slotted CDMA System (Slotted CDMA 환경에서 멀티미디어 트래픽의 QoS 보장을 위한 MAC 프로토콜)

  • 동정식;이형우;조충호
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.8B
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    • pp.707-715
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    • 2003
  • In this paper, we propose a new MAC(Medium Access Control) protocol using Movable-boundary, which tries to guarantee Qos for multimedia traffic in the slotted CDMA system. In this scheme, the traffic scheduler assigns channel resource according to the packet priority per service class and adapts the Movable-boundary concept in which the minimum resource is assigned to each traffic class; the remaining resource if it is available can be assigned dynamically according to the temporal demand of other traffic classes. For performance analysis, we performed computer simulations to obtain throughput and packet loss rate and compared the results with Fixed-boundary system. We observed that the error rate of voice traffic could be maintained below a prescribed value while bursty traffic such as video source shares the same channel. In comparison with Fixed-boundary scheme, our protocol exhibits better throughput and packet loss rate performance.

Cooperative Communication Scheme Based on channel Characteristic for Underwater Sensor Networks (수중 센서 네트워크를 위한 채널 특성기반의 협력 통신 기법)

  • Ji, Yong-Joo;Choi, Hak-Hui;Lee, Hye-Min;Kim, Dong-Seong
    • Journal of the Institute of Electronics and Information Engineers
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    • v.53 no.6
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    • pp.21-28
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    • 2016
  • This paper presents a cooperative transmission scheme for underwater acoustic sensor networks to improve packet transmission rate and reduce energy consumption. Source node transmits duplicated information relayed by distributed antennas called a virtual antenna array. Destination node combines that information to reduce packet error rate. The suggested cooperative scheme enhances the reliability by providing high diversity gains through intermediate relay nodes to overcome the distinct characteristics of the underwater channel, such as high transmission loss, propagation delay, and ambient noises. It is suggested that the algorithm select destinations and potential relays from a set of neighboring nodes that utilize distance cost, the residual energy of each node and local measurement of the channel conditions into calculation. Simulation results show that the proposed scheme reduces average energy consumption, response time, and increases packet delivery ratio compared with the SPF(Shortest Path First) and non-cooperative scheme using OPNET Moduler.

ACCB- Adaptive Congestion Control with backoff Algorithm for CoAP

  • Deshmukh, Sneha;Raisinghani, Vijay T.
    • International Journal of Computer Science & Network Security
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    • v.22 no.10
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    • pp.191-200
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    • 2022
  • Constrained Application Protocol (CoAP) is a standardized protocol by the Internet Engineering Task Force (IETF) for the Internet of things (IoT). IoT devices have limited computation power, memory, and connectivity capabilities. One of the significant problems in IoT networks is congestion control. The CoAP standard has an exponential backoff congestion control mechanism, which may not be adequate for all IoT applications. Each IoT application would have different characteristics, requiring a novel algorithm to handle congestion in the IoT network. Unnecessary retransmissions, and packet collisions, caused due to lossy links and higher packet error rates, lead to congestion in the IoT network. This paper presents an adaptive congestion control protocol for CoAP, Adaptive Congestion Control with a Backoff algorithm (ACCB). AACB is an extension to our earlier protocol AdCoCoA. The proposed algorithm estimates RTT, RTTVAR, and RTO using dynamic factors instead of fixed values. Also, the backoff mechanism has dynamic factors to estimate the RTO value on retransmissions. This dynamic adaptation helps to improve CoAP performance and reduce retransmissions. The results show ACCB has significantly higher goodput (49.5%, 436.5%, 312.7%), packet delivery ratio (10.1%, 56%, 23.3%), and transmission rate (37.7%, 265%, 175.3%); compare to CoAP, CoCoA+ and AdCoCoA respectively in linear scenario. The results show ACCB has significantly higher goodput (60.5%, 482%,202.1%), packet delivery ratio (7.6%, 60.6%, 26%), and transmission rate (40.9%, 284%, 146.45%); compare to CoAP, CoCoA+ and AdCoCoA respectively in random walk scenario. ACCB has similar retransmission index compare to CoAp, CoCoA+ and AdCoCoA respectively in both the scenarios.

Performance Analysis of WAP Packet Considering WTP SAR Algorithm and RLP in Wireless CDMA Network (무선 CDMA 망에서 WTP SAR 알고리즘과 RLP를 고려한 WAP 패킷의 성능 분석)

  • Moon, Il-Young;Roh, Jae-Sung;Cho, Sung-Joon
    • Journal of Advanced Navigation Technology
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    • v.6 no.1
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    • pp.69-76
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    • 2002
  • With the growth of data communication service by mobile devices, WAP is proposed to efficiently access the Internet contents by user request through wireless condition that has a high error rate and mobility. But, because a transmission speed of WAP is limited, it takes many times to transmit and to receive the data. This paper has studied the WAP packet transmission time using WTP-SAR algorithm. As a method that is to improve transfer capability of WAP, using SAR function in WTP, total message down from upper layer has been fragmented and packet is transmitted through RLP frame time slot. Then, we have analyzed the transmission time of WAP packet with variable RLP layer size on the wireless CDMA network for next generation systems. From the results, we could obtain the WAP packet transmission time and optimal WTP packet size.

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