• The Journal of Information Technology
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• v.9 no.4
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• pp.57-63
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• 2006

The intelligent elder brother groove network robot service the new broadband presence line is conversionce application service. Robot control data transmission hazard UDP packet of the remote control data which stands duplication necessary to transmit. TCP the error ratio to be high qualitative recording transmission pattern it of the transmission unit and is irregular the distance is distant to show and also the transmission lag is visible increases. The recording packet drop whose UDP degree error ratio will be high is frequent and does not arrive packet little by little increases is a possibility of knowing in the transmission unit. The technique which it proposes with traffic pattern of the transmission unit is visible the transfer characteristic of the same shape from 1% packet error ratio degree. The effective transmission technique of the robot control data which puts a base in UDP protocols was proposed from the present paper. Following research it leads and it follows the duplication transmission number of time in error rate of radio link and it was thought all that controls petty the research of the mechanism which progresses is necessary.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.4
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• pp.581-587
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• 2021

Real-time video streaming scheme for multimedia content delivery and remote conference services is one of technologies that are significantly sensitive to data transmission delay. Recently, because of COVID-19, real-time video streaming contents for the services are significantly increased such as personal broadcasting and remote school class. In order to support the services, there is a growing emphasis on low transmission delay and secure content delivery, respectively. Therefore, our research proposed a packet aggregation algorithm to reduce the transmission delay of selectively encrypted video transmission for real-time video streaming services. Through the application of the proposed algorithm, the selectively encrypted video framework can control the amount of MPEG-2 TS packets for low latency transmission with a consideration of packet priorities. Evaluation results on testbed show that the application of the proposed algorithm to the video framework can reduce approximately 11% of the transmission delay for high and low resolution video.

• IEMEK Journal of Embedded Systems and Applications
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• v.6 no.3
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• pp.149-156
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• 2011

Transmission delay and packet loss induced by a communication network can degrade the control performance and, even make the system unstable. This paper presents a method for compensating transmission delay and packet loss in a networked control system for unmanned underwater vehicle. The proposed method is based on Lagrange interpolation in order to satisfy the requirements of simplicity and model-independency. In this work, the lost/delayed data are estimated in real time by only using the past data without requiring any mathematical model of the controlled system. Consequently, the proposed method can be implemented independent of the controlled system, and also it can achieve fast and accurate compensation performance. The performance of the proposed technique is evaluated by numerical simulations with an unmanned underwater vehicle.

• International Journal of Computer Science & Network Security
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• v.22 no.3
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• pp.1-8
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• 2022

Supporting real-time flows with delay and throughput constraints is an important challenge for future wireless networks. In this paper, we develop an optimal scheduling scheme to optimally choose the packets to transmit. The optimal transmission strategy is based on an observable Markov decision process. The novelty of the work focuses on a priority-based probabilistic packet scheduling strategy for efficient packet transmission. This helps in providing guaranteed services to real time traffic in Heterogeneous Wireless Networks. The proposed scheduling mechanism is able to optimize the desired performance. The proposed scheduler improves the overall end-to-end delay, decreases the packet loss ratio, and reduces blocking probability even in the case of congested network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.8C
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• pp.810-823
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• 2005

The transmission of duplicate packets provides a great loss-resilience without undue time-delay in the video transmission over packet loss networks. But this method generally deteriorates the problem of traffic congestion because of the increased bit-rate required for duplicate transmission. In this paper, we propose an efficient packetization and duplicate transmission of video packets. The proposed method transmits only the video signal with high priority for each video macroblock that is quite small in volume but very important for the reconstruction of the video. The proposed method significantly reduces the required bit-rate for duplicate transmission. An efficient packetization method is also proposed to reduce additional packet overhead which is required for transmitting the duplicate data. The duplicated high priority data of the Previous video slice is transmitted as a Piggyback to the data Packet of the current video slice. It is shown by simulations that the proposed method remarkably improves the packet loss-resilience for video transmission only with small increase of redundant duplicated data for each slice.

• Journal of Advanced Navigation Technology
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• v.8 no.1
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• pp.65-72
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• 2004

A basic access method using for IEEE 802.11a wireless LANs is the DCF method that is based on the CSMA/CA. But, Since IEEE 802.11 MAC layer uses original backoff algorithm (Exponential backoff method), when collision occurs, the size of contention windows increases the double size. Hence, packet transmission delay time increases and efficiency is decreased by original backoff scheme. In this paper, we have analyzed TCP packet transmission time of IEEE 802.11 MAC DCF protocol for wireless LANs using a proposed enhanced backoff algorithm. From the results, in OFDM/quadrature phase shift keying channel (QPSK), we can achieve that the transmission time in wireless channel decreases as the TCP packet size increases and based on the data collected, we can infer the correlation between TCP packet size and total message transmission time, allowing for an inference of the optimal packet size in the TCP layer.

• Journal of Information Technology Applications and Management
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• v.23 no.2
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• pp.61-80
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• 2016

The wireless mesh network (WMN) is a next-generation technology for data networking that has the advantage in cost and the flexibility in its construction because of not requiring the infra-structure such as the ethernet. This paper focuses on the optimal link scheduling problem under the wireless mesh network to effectuate real-time streaming by using the constraint programming. In particular, Under the limitation of half-duplex transmission in wireless nodes, this paper proposes a solution method to minimize the makespan in scheduling packet transmission from wireless nodes to the gateway in a WMN with no packet transmission conflicts due to the half-duplex transmission. It discusses the conflicts in packet transmission and deduces the condition of feasible schedules, which defines the model for the constraint programming. Finally it comparatively shows and discusses the results using two constraint programming solvers, Gecode and the IBM ILOG CP solver.

• 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.

• 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.

• 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.