• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• Journal of Korea Multimedia Society
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• v.4 no.4
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• pp.349-355
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• 2001

In this paper, we newly propose a performance enhanced CSMA/CD MAC(Medium Access Control) protocol for the Ethernet which changes its operation mode dynamically according to the network status, not fixed it as one of p-persistent mode and non-persistent mode. Dynamic mode change occurs independently on each node, and uses the consecutive success count and the fail count of the frame transmission. The simulation result shows that the dynamic mode change maintains the enhanced network utilization and transmission delay characteristics. Also we show the implementation simplicity of our MAC protocol through its conceptual design using the Ethernet commercial chip as it stands.

• Journal of the Institute of Convergence Signal Processing
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• v.3 no.1
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• pp.21-26
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• 2002

Video stream invoke a variety of traffic with the structure of compression algorithm and image complexity. Thus, it is difficult to allocate the resource on the both sides of sender and receiver, and playout on the Internet such as a packet switched network. Thus, in this paper we proposed video stream smoothing using multistream for the effective transmission of video stream. This method specifies the type of LDU(logical data unit) according to the type of original stream, and then makes a large number of streams as a fixed size, and transfers them. So, the proposed method can reduce the buffering time which occurs during the process of the smoothing and prefetch be robust to the jitter on network, as well. Consequently, it has the effective transmission characteristics of fully utilizing the clients bandwidth.

• Proceedings of the KAIS Fall Conference
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• 2004.11a
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• pp.199-202
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• 2004

본 논문에서는 유선망과 무선망이 혼재된 네트워크 환경에서 유선망에서 발생한 혼잡 상황을 송신측에 빨리 알려서 손실된 패킷에 대한 신속한 fast retransmission을 가능하도록 하는 기법을 제안한다. 제안된 기법은 유선망과 무선망의 경계에 있는 기지국에 구현되는데, 유선망으로부터 순서가 유지되지 않은 패킷이 3 개 연속해서 오게 되면 수신부로부터 3 개의 중복 ACK 세그먼트가 오기를 기다리지 말고 즉시 송신측에 3 개의 중복 ACK를 인위적으로 만들어 전송함으로써 송신측이 망의 혼잡 상황을 빨리 파악하여 대처할 수 있도록 한다. 특히 무선망 환경은 유선망에 비해 상대적으로 에러가 많이 발생하는 환경이므로 유선망의 혼잡 상황을 무선망의 수신부 측이 중복 ACK를 통해 알도록 하는 것은 상당한 지연을 야기 할 수 있으므로 제시된 기법은 이러한 환경에서 효과적으로 사용될 수 있다. 제안된 기법의 성능 향상 효과를 NS-2를 통한 시뮬레이션을 통해 확인하였다.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.2
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• pp.16-27
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• 2017

IEEE 802.15.6 standard, a Wireless Body Area Network, aims to transfer not only medical data but also non-medical data, such as physical activity, streaming, multimedia game, living information, and entertainment. Services which transfer those data have very various data rates, intervals and frequencies of continuous access to a medium. Therefore, an efficient anti-collision operations and medium assigning operation have to be carried out when multiple nodes with different data rates are accessing shared medium. IEEE 802.15.6 standard for CSMA/CA medium access control method distributes access to the shared medium, transmits a control packet to avoid collision and checks status of the channel. This method is energy inefficient and causes overhead. These disadvantages conflict with the low power, low cost calculation requirement of wireless body area network, shall minimize such overhead for efficient wireless body area network operations. Therefore, in this paper, we propose a medium access scheduling scheme, which adjusts the time interval for accessing to the shared transmission medium according to the amount of data for generating respective sensor node, and a priority control algorithm, which temporarily adjusts the priority of the sensor node that causes transmission concession due to the data priority until next successful transmission to ensure fairness.

• Journal of Korea Multimedia Society
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• v.5 no.6
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• pp.697-703
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• 2002

In this paper, we propose the adaptive multi stream transmission technique based on SPIHT video signal for the highest quality service over the current Internet that does not guarantee QoS. In addition to the reliable transmission of the video stream over the asynchronous packet network, the proposed approach provides the transmission using the adaptive frame pattern control and multi steam over the TCP for continuous replay. The adaptive frame pattern control makes the transmission date scalable in accordance with the client's buffer status. Apart from this, the multi stream transmission improves the efficiency of video stream, and is robust to the network jitter problem, and maximally utilizes the bandwidth of the client's. As a result of the experiment, the DR(delay ratio) in the proposed adaptive multi-stream transmission is more close to zero than in the existing signal stream transmission, which enables the best-efforts service to be implemented.

• Journal of KIISE:Computer Systems and Theory
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• v.30 no.1
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• pp.22-32
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• 2003

Many of multimedia applications in distributed environments generate the packets which have the real-time characteristics for continuous audio/video data and transmit them according to the teal-time task scheduling theories. In this paper, we model the traffic for continuous media in the distributed multimedia applications based on the high-bandwidth networks and introduce the PDMA algorithm which is the hard real-time task scheduling theory for guaranteeing QoS requested by the clients. Furthermore, we propose the admission control to control the new request not to interfere the current services for maintaining the high quality of services of the applications. Since the proposed admission control is sufficient for the PDMA algorithm, the PDMA algorithm is always able to find the feasible schedule for the set of messages which satisfies it. Therefore, if the set of messages including the new request to generate the new traffic. Otherwise, it rejects the new request. In final, we present the simulation results for showing that the scheduling with the proposed admission control is of practical use.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.1 no.2
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• pp.145-158
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• 1997

Method for synchronizing multimedia data is needed to support continuous transmission of multimedia data through a network in a bounded time and it also required for supporting continuous presentation of multimedia data with the required norminal playout rate in distributed network environments. This paper describes a new synchronization method for supporting delay-sensitive multimedia Presentation without degration of Quality of services of multimedia application. It mainly aims to support both intermedia and intermedia synchronization by absorbing network variations which may cause skew or jitter. In order to remove asynchonization problems, we make use of logical time system, dynamic buffer control method, and adjusting synchronization intervals based on the quality of services of a multimedia. It might be more suitable for working on distribute[1 multimedia systems where the network delay variation is changed from time to time and no global clock is supported. And it also can effectively reduce the amount of buffer requirements needed for transfering multimedia data between source and destination system by adjusting synchronization intervals with acceptable packet delay limits and packet loss rates.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10A
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• pp.965-973
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• 2007

We analyzed that the loss of data in TCP protocol based wireless networks caused by overlapped responses in bi-directional nodes that were resulted in out of the data sequence. This loss can be prevented by using revised TCP rate control algorithm and the performance of throughput can also be improved. The rate control algorithm is applied when the congestion happens between nodes while traffic packets are retransmitting in TCP bandwidth. In addition to applying the rate control algorithm, we determine the number of system clients in bandwidth and the average of pausing time between transmitting serial files to produce a competitive level so that an efficient performance of rapid retransmitting for the loss of multi-packets. This paper discusses the improvement of congestion control in that the decrease of the loss, firstly, as ensuring an efficient connection rate and, secondly, as using sliding window flow control.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.