• The Journal of the Korea Contents Association
• /
• v.12 no.4
• /
• pp.69-75
• /
• 2012

In this paper, to provide various services of QoS, and moreover applying traffic ratio to CAC(Call Admission Control) algorithm tested how long average data rate and the average packet delay time. When CAC algorithm is not applied, traffic mixture ratio is 1:1:4:4, the FTP Service 0.4, web services 0.6, streaming service 0.7, the packet delay requirements are not satisfied. On the other hand CAC Algorithm is applied, all the service of packet delay are satisfied with arrival rate. Therefore, we can make sure that applying of CAC of traffic control WWW, FTP, Video, VoIP can guarantee the various services of QoS.

• Journal of Advanced Navigation Technology
• /
• v.13 no.4
• /
• pp.523-531
• /
• 2009

In this study, we propose a joint source/channel rate control algorithm for video encoder targeting packet erasure channel. Based on the buffer constraints of video communication systems, encoding rate constraint is presented. After defining source distortion models for coded and skipped video frames and a channel distortion model for packet errors and their propagation, an average distortion model of received video is proposed for a given encoding window. Finally, we define an optimization problem to minimize the average distortion for given channel rates and packet loss rates by controlling spatio-temporal parameters of source video and FEC block sizes. Then, we propose a window-based algorithm to solve the problem in real-time.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.11B
• /
• pp.953-959
• /
• 2006

This paper provides optimum TCP/IP packet size that maximizes the throughput efficiency of ATM layer as a function of TCP/IP packet length for several values of channel BER over wireless ATM LAN links applying data link error control schemes to reduce error problems encountered in using wireless links. For TCP/IP delay-insensitive traffc requiring reliable delivery, it is necessary to adopt data link layer ARQ protocol. So ARQ error control schemes considered in this paper include GBN ARQ, SR ARQ and type-I Hybrid ARQ, which ARQ is needed, but FEC can be used to reduce the number of retransmissions. Especially adaptive type-I Hybrid ARQ scheme is necessary for a variable channel condition to make the physical layer as SONET-like as possible.

• Journal of information and communication convergence engineering
• /
• v.1 no.4
• /
• pp.194-198
• /
• 2003

Recently, the use of TCP/IP protocols over wireless LANs poses significant problems. In this paper, we have analyzed transmission control protocol (TCP) packet transmission time for mobile IP over wireless local area networks (LANs) using a proposed a new random backoff scheme. We call it as a proxy backoff scheme. It is considered the transmission time of TCP packet on the orthogonal frequency division multiplexing (OFDM) in additive white gaussian noise (AWGN) channel. From the results, a proposed proxy backoff scheme produces a better performance than an original random backoff in mobile IP over wireless LANs environment. Also, in OFDM/quadrature phase shift keying (QPSK) medium access control (MAC), we have obtained that the transmission time in wireless channel decreases as the TCP packet size increases.

• Proceedings of the KIEE Conference
• /
• 2006.04a
• /
• pp.105-107
• /
• 2006

As internet streaming data increase, transport protocol such as TCP, TGP-Friendly is important to study control transmission rate and share of Internet bandwidth. In this paper, we propose a TCP-Friendly protocol using Neural Network for media delivery over wired Internet which has various traffic size(PTFRC). PTFRC can effectively send streaming data when occur congestion and predict one-step ahead round trip time and packet loss rate. A multi-layer perceptron structure is used as the prediction model, and the Levenberg-Marquardt algorithm is used as a traning algorithm. The performance of the PTFRC was evaluated by the share of Bandwidth and packet loss rate with various protocols.

• Proceedings of the KIPE Conference
• /
• 2002.07a
• /
• pp.635-638
• /
• 2002

In this paper, the connection device of RS232C and TCP/IP implementation using ARM processor and LINUX is proposed. Data interaction flash memory the multiple serial ports are transferred to ARM processor and the data are processed and formed into data packet for transfer via internet protocol. Packet flash memory Internet is decoded to extract the serial port data. The serial ports supports RS232C asynchronous protocol communication and control program is developed in GNU-C and installed in the on-board memory for packet conversion and control. The research result can be applied to terminal server, printer server and multiple serial ports equipments.

• Proceedings of the KIEE Conference
• /
• 2000.07d
• /
• pp.3028-3030
• /
• 2000

In general. a speech coding is dedicated to the compression performance or the speech quality. But. the speech coding in this paper is focused on the performance of flexible transmission to the, network speed. For this. the subbanding coding is needed. which is used the wavelet packet concept in the signal analysis. The extraction of each frequency-band is difficult to general signal analysis methods, after coding each band, the reconstruction of these is also a difficult problem. But. with the wavelet packet concept(perfect reconstruction) and its fast computation algorithm. the extraction of each band and the reconstruction are more natural. Also, this paper describes a direct solution of the voice transmission on network and implement this algorithm at the TCP/IP network environment of PC.

• Journal of information and communication convergence engineering
• /
• v.17 no.2
• /
• pp.97-104
• /
• 2019

Distributed mobility management (DMM) does not use a centralized device. Its mobility functions are distributed among routers; therefore, the mobility services are not limited to the performance and reliability of specific mobility management equipment. The DMM scheme has been studied as a partially distributed architecture, which distributes only a packet delivery domain in combination with the software defined network (SDN) technology that separates the packet delivery and control areas. Particularly, a separated control area is advantageous in introducing a new service, thereby optimizing the network by recognizing the entire network situation and taking an optimal decision. The SDN-based mobility management scheme is studied as a method to optimize the packet delivery path whenever a mobile node moves; however, it results in excessive signaling processing cost. To reduce the high signaling cost, we propose a hybrid distributed mobility management method and analyze its performance mathematically.

• Proceedings of the Korea Contents Association Conference
• /
• 2006.11a
• /
• pp.75-79
• /
• 2006

In a VoIP(Voice over IP) system which support the realtime speech service, speech quality is deteriorated by the delay, the jitter, the loss, and the reversed packet order. In this thesis, APBC for receiver site is proposed, which compensate the jitter by the adaptive playout algorithm and conceal the packet loss, and align the packet order. Also, a VoIP application system is implemented, and the performance of APBC is verified on the implemented system by measuring the processing speed and the speech quality. From the result, processing speed is 257$\mu$sec, which is fast enough to deal with packet being received in realtime. Also, the speech quality by MOS(Mean Opinion Score) is improved as 18 percent compared with algorithm of fixed playout delay.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.4 no.3
• /
• pp.341-357
• /
• 2010

Forward Error Correction (FEC) techniques have been adopted to overcome packet losses and to improve the quality of video delivery. The efficiency of the FEC has been significantly compromised, however, due to the characteristics of the wireless channel such as burst packet loss, channel fluctuation and lack of Quality of Service (QoS) support. We propose herein an Adaptive Cross-layer FEC mechanism (ACFEC) to enhance the quality of video streaming over 802.11 WLANs. Under the conventional approaches, FEC functions are implemented on the application layer, and required feedback information to calculate redundancy rates. Our proposed ACFEC mechanism, however, leverages the functionalities of different network layers. The Automatic Repeat reQuest (ARQ) function on the Media Access Control (MAC) layer can detect packet losses. Through cooperation with the User Datagram Protocol (UDP), the redundancy rates are adaptively controlled based on the packet loss information. The experiment results demonstrate that the ACFEC mechanism is able to adaptively adjust and control the redundancy rates and, thereby, to overcome both of temporary and persistent channel fluctuations. Consequently, the proposed mechanism, under various network conditions, performs better in recovery than the conventional methods, while generating a much less volume of redundant traffic.