• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.652-657
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• 2004

As the real-time multimedia applications through Internet increase, the bandwidth available to TCP connections is oppressed by the UDP traffic, result in the performance of overall system is extremely deteriorated. Therefore, developing a new transmission protocol is necessary. The TCP-friendly algorithm is an example meeting this necessity. The TCP-friendly (TFRC) is an UDP-based protocol that controls the transmission rate based on the available round transmission time (RTT) and the packet loss rate (PLR). In the data transmission processing, transmission rate is determined based on the conditions of the previous transmission period. If the one-step ahead predicted values of the control factors are available, the performance will be improved significantly. This paper proposes a prediction model of transmission rate control factors that will be used for the transmission rate control, which improves the performance of the networks. The model developed through this research is predicting one-step ahead variables of RTT and PLR. A multiplayer perceptron neural network is used as the prediction model and Levenberg-Marquardt algorithm is used for the training. The values of RTT and PLR were collected using TFRC protocol in the real system. The obtained prediction model is validated using new data set and the results show that the obtained model predicts the factors accurately.

• Journal of IKEEE
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• v.22 no.3
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• pp.651-657
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• 2018

IEEE 802.11 wireless LANs support 20, 40, 80 and 160MHz bandwidth transmission. In general, the data rate increases as the transmission bandwidth increases. However, the transmission power spectral density decreases, which may lead to increasing packet errors and retransmissions. In this paper, we derive a mathematical model of energy consumption with consideration of various factors such as transmission bandwidth, packet error rate and data size. Based on the model, we design a scheme to adapt a transmission bandwidth for each frame transmission. The scheme estimates packet error rates for different bandwidth cases, updates the table of energy consumption and selects the best bandwidth for the next transmission. The simulation study with VoIP traffic shows the energy consumption of the scheme under various environments.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3E
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• pp.99-108
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• 2009

In this paper, we suggest criteria for objective measurement of speech quality in mobile VoIP (Voice over Internet Protocol) services over wireless mobile internet such as mobile WiMAX networks. This is the case that voice communication service is available under other networks. When mobile VoIP service users in the mobile internet network based on packet call up PSTN and mobile network users, but there have not been relevant quality indexes and quality standards for evaluating speech quality of mobile VoIP. In addition, there are many factors influencing on the speech quality in packet network. Especially, if the degraded speech with packet loss transfers to the other network users through the handover, voice communication quality is significantly deteriorated by the transformation of speech codecs. In this paper, we eventually adopt the Gilbert-Elliot channel model to characterize packet network and assess the voice quality through the objective speech quality method of ITU-T P. 862. 1 MOS-LQO for the various call scenario from mobile VoIP service user to PSTN and mobile network users under various packet loss rates in the transmission channel environments. Our simulation results show that transformation of speech codecs results in the degraded speech quality for different transmission channel environments when mobile VoIP service users call up PSTN and mobile network users.

• IEMEK Journal of Embedded Systems and Applications
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• v.1 no.1
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• pp.24-30
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• 2006

With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In these days, many papers describe about wireless-TCP which has suggested how to avoid congestion control when packet loss over the wireless network. In this paper, an enhancement scheme is proposed by modifying SNOOP scheme. To enhance the original SNOOP scheme, CPC(Consecutive Packet Control) and ZWSC(Zero Window Size Control) are added. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compares to SNOOP by knowing the cause of packet loss at the base station.

• Journal of Digital Contents Society
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• v.10 no.2
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• pp.279-289
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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. In this article, we suggest the newly improved algorithm using two parameters, the local retransmission time value and the local retransmission critical value to the BS based on the Snoop. This technique adjusts the base stations local retransmission timer effectively according to the wireless link status to recover the wireless packet loss rapidly. We checked that as a result of the suggested algorithm through various simulations, A-Snoop protocol improve more the wireless TCP transmission rate by recovering the packet loss effectively in the wireless link that the continuous packet loss occur than the Snoop protocol.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1483-1486
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• 2002

Multicast protocols are efficient methods of group communication, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The purpose of this dissertation is to find a method to utilize sewer routers to form multicasts that can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM method has been used. Three packets per TCP transmission control window size are used for transport and an ACK is used for flow control. A CBR and a SRM is used for UDP traffic control. Divided on whether a UDP multicast packet and TCP unicast packet is used simultaneously or only a UDP multicast packet transport is used, the multicast receiver with the longest delay is measured on the number of packets and its data receiving rate. It can be seen that the UDP packet and the TCP's IP packet can be simultaneously used in a server router.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.7
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• pp.923-929
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• 2021

TCP communication and packet communication require transmission control technology to ensure high quality and high reliability. However, in the case of real-time data transmission, an inefficient transmission problem occurs. In order to overcome this problem and transmit the packet reliability, in general, early congestion control using the buffer level as an index was used. Control of the congestion control point and the cancellation point is delayed because the point at which congestion is controlled is based on the buffer level. Therefore, in this paper, not only the buffer level indicator, but also the ideal buffer level, which determines the packet discard probability, is classified so that the transmission rate and buffer level that measure network congestion are close to the level above the optimal setting. As a result, it was shown that the average buffer level can be directly controlled by maintaining the average buffer level by the ideal buffer level set in the experiment to prove the proposed method.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.208-212
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• 2003

2.4GHz wireless LAN has come into wide use at present according as the use of wireless LAN limited to special fields like network of enterprises or universities has been made gradual popularization and the need for it is gathering strength. Therefore, in this paper, transmission devices and structures of it is investigated and optimum transmission level of signals is researched through experiment about characteristics and rate of transmission of Packet signals in various User environments.

• Journal of KIISE
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• v.44 no.10
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• pp.1062-1070
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• 2017

Real-time transmission studies in wireless sensor networks propose a mechanism that exploits a node that has a higher delivery speed than the desired delivery speed in order to satisfy real-time requirement. The desired delivery speed cannot guarantee real-time transmission in a congested area in which none of the nodes satisfy the requirement in one hop because the desired delivery speed is fixed until the packet reaches the sink. The feature of this mechanism means that the packet delivery speed increases more than the desired delivery speed as the packet approaches closer to the sink node. That is, the packet can reach the sink node earlier than the desired time. This paper proposes an improved real-time transmission by controlling the delivery speed using the temporal gain which occurs on the packet delivery process. Using the received data from a previous node, a sending node calculates the speed to select the next delivery node. The node then sends a packet to a node that has a higher delivery speed than the recalculated speed. Simulation results show that the proposed mechanism in terms of the real-time transmission success ratio is superior to the existing mechanisms.