• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7A
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• pp.485-492
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• 2003

This paper proposes a transmission probability control scheme for guaranteeing fair packet transmissions in CDMA slotted ALOHA system. In CDMA slotted ALOHA system, the packets transmitted in the same slot act as multiple access interference, so that unsuccessful packet transmissions are caused entirely by multiple access interference. Therefore, in order to maximize the system throughput, the number of simultaneously transmitted packets should be kept at a proper level. In the proposed scheme, the base station calculates the packet transmission probability of mobile stations in the next slot according to the offered load and then broadcasts this probability to all the mobile stations. Mobile stations, which have a packet to transmit, attempt to transmit packet with the received probability. Simulation results show that the proposed scheme can offer better system throughput and average delay than the conventional scheme, and guarantee a good fairness among all mobile stations regardless of the offered load.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.6
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• pp.689-694
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• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.7
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• pp.553-563
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• 2013

When the IEEE 802.11 wireless LAN-based broadcasting services, we use broadcast packets to broadcast multimedia contents to a large number of users using limited wireless resources. However, broadcast transmission is difficult to recover the loss packets compared with unicast transmission. Therefore, analysis of packet loss characteristics is required to perform efficient packet recovery. The packet loss in wireless transmissions is often bursty with high loss data rate. Even if loss patterns have the same average packet loss, they are different in the recovery rate of random loss and burst loss depending on the nature. Therefore, the analysis and research of the nature of the loss are needed to recover loss packets considering bursty characteristics. In this paper, we experimented Wi-Fi broadcast transmission according to transmission rate and analyzed bursty characteristics of loss patterns using 4-state markov model.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.8
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• pp.77-85
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• 1998

This paper proposes progressive image transmission(PIT) methods based on the wavelet packet transform, in which quantizers are optimized at each stage for the given bit rate. Scalar and vector quantizers are used and the performance of each quantizer is compared. After quantization, selected subbands are ordered by their priority for transmission. Subjective quality of the reconsetructed image is improved by human visual system (HVS) weighting.

• Journal of information and communication convergence engineering
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• v.5 no.3
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• pp.181-184
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• 2007

In this paper, a CSSMA/AI MAC protocol in packet CDMA network is presented. The main features of this protocol are the code status sensing and reservation for reducing the packet collision. The base station broadcasts the code status on a frame-by-frame basis just before the beginning of each preamble transmission, and the mobile station transmits a preamble for reserving a code based on the received code status. After having transmitted the preamble, the mobile station listens to the downlink of the selected code and waits the base station reply. If this reply indicates that the code has been correctly acquired, it continues the packet transmission for the rest of the frame. If there are other packets waiting for transmission, the base station broadcasts the status of the code as reserved, and the mobile station transmits a packet on a reserved code for the successive frames.

• Speech Sciences
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• v.11 no.4
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• pp.67-73
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• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10C
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• pp.950-958
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• 2007

This paper presents an efficient packet loss resilient scheme for real-time video transmission over the Internet. By analyzing the temporal and spatial dependencies in inter- and intra-frames, we assign forward error correction codes (FEC) across video packets for minimizing the effect of error concealment and error propagation from packet loss. To achieve optimal allocation of FEC codes, we formulate the effect of packet loss on video quality degradation as packet distortion model. Then we propose an unequal FEC assignment scheme with low complexity based on packet correction rate, which uses the packet distortion model and includes channel status information. Simulation results show that the proposed FEC assignment scheme gives substantial improvement for the received video quality in packet lossy networks. Furthermore the proposed scheme achieves relatively smaller degradation of video quality with higher packet loss rates.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.15 no.4
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• pp.2382-2392
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• 2014

In this paper, a new data frame structure for wavelength routing in optical ring access network(ORAN) is designed and the functions of each field in the frame are defined. In addition, the IP packet transmission to the network layer corresponding to the newly proposed structure of the protocol stack is newly presented. Tha data transmission protocol design techniques of ORAN was proposed. IP packet and broadcasting packet transmission within the sub-network, and the IP packet transmission to other sub-network was shown through the process illustration. In the process, the encapsulation and framing process of the wavelength information to routing has been described in detail. And each step takes place in a packet transfer process is demonstrated. A ring type optical access network protocol is not yet research field. ORAN data transfer protocol to send/receive module structure of a subscriber node and a control node of the two rings which analyzed the results are shown. The high-speed Internet solution is proposed that by using wavelength routing the packet transport protocol for ORAN is designed.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.923-928
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• 2003

This paper proposes a bottleneck link bandwidth measurement algorithm for reducing packet transmission delay within the grid network. There are two methods for measuring bottleneck link bandwidth：Packet Pair algorithm and Paced Probes algorithm. They measure bottleneck link bandwidth using the difference in arrival times of two paced probe packets of the same size traveling from the same source to destination. In addition, they reduce the influences of cross traffic by pacer packet. But there are some problems on these algorithms：it's not possible to know where bottleneck link occurred because they only focus on measuring the smallest link bandwidth along the path without considering bandwidth of every link on the path. So hop-by-hop based bottleneck link bandwidth measurement algorithm can be used for reducing packet transmission delay on grid network. Timestamp option was used on the paced probe packet for the link level measurement of bottleneck bandwidth. And the reducing of packet transmission delay was simulated by the solving a bottleneck link. The algorithm suggested in this paper can contribute to data transmission ensuring FTP and realtime QoS by detecting bandwidth and the location where bottleneck link occurred.

• Journal of the Korea Society of Computer and Information
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• v.16 no.2
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• pp.131-140
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• 2011

Smart packet agent is an application that is proposed to provide routing protocol and service module in ubiquitous network environment. However, it uses multi-hop broadcast, thus it causes increasing network traffic, low-speed data transmission, and the unnecessary joining nodes. In this paper, a transmit technique that uses unicast-based multi-hop to have lower network traffic and faster transmission time than the multi-hop broadcast technique. In our scheme, u-Zone Master establishes temporary routing paths by calculating moving nodes' hop-counter. Therefore, it reduces smart packet agent's network traffic and retransmission rate. Besides, this paper proposed an UDP transmission that bases on sliding window. Hence, the Stop & Wait transmission speed is improved. The results, which are taken by analyzing performance prove that the proposed scheme has better performance.