• Journal of Communications and Networks
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• 제7권4호
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• pp.408-416
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• 2005

Quality of service (QoS) provision is an important and indispensable function for multi-service wireless networks. In this paper, we present a new scheduling/admission control frame­work, including an efficient rate-guaranteed opportunistic scheduling (ROS) scheme and a coordinated admission control (ROS­CAC) policy to support statistic QoS guarantee in multi-service wireless networks. Based on our proposed mathematical model, we derive the probability distribution function (PDF) of queue length under ROS and deduce the packet loss rate (PLR) for individual flows. The new admission control policy makes admission decision for a new incoming flow to ensure that the PLR requirements of all flows (including the new flow) are satisfied. The numerical results based on ns-2 simulations demonstrate the effectiveness of the new joint packet scheduling/admission control framework.

• 한국게임학회 논문지
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• 제2권1호
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• pp.23-29
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• 2002

Multicast protocols are efficient methods of group communication such as video conference, Internet broadcasting and On-Line Game, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The Purpose or this Paper is to find a method to utilize multicast routers can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM(Scalable Reliable Multicast)method has been used. Three packets per TCP transmission control window site are used for transport and an ACK is used for flow control. A CBR(Constant Bit Rate) 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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(1)
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• pp.45-48
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• 2002

This paper presents a QoS-guaranteed traffic control system which supports QoS of realtime packet transmission for the multimedia communication. The traffic control system presented in this paper applies the integrated service model and provides QoS o(packet transmission by means of determining the packet transmission rate with the policy of network manager and the optimal resource allocation according to the end-to-end traffic load. It also provides QoS for the realtime packet transmission through the AWF2Q+ Scheduling algorithm and per-class queuing method.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2001년도 ICCAS
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• pp.40.1-40
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• 2001

In this paper, control theoretic approaches are proposed to guarantee QoS (Quality of Series) such as packet delay and packet loss of real-time traffic in high-speed communication network. Characteristics of variable rate real-time traÆc in communication networks are described. The mathematical model describing networks including source and destination nodes are suggested. By a traffic control mechanism, it is shown that worst-case end-to-end transfer delay of traffic can be controlled and packet loss can be prevented. The simulation shows results of delay control and buer level control to raise QoS in realtime traffic.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제5권6호
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• pp.1147-1165
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• 2011

We present a simple and cost effective rate control scheme for streaming video over a wireless channel by using the information of mobile devices' buffer level. To prevent buffer fullness and emptiness at receivers, the server should be able to adjust sending rate according to receivers' buffer status. We propose methods to adjust sending rate based on the buffer level and discrete derivative of the buffer occupancy. To be compatible with existing network protocols, we provide methods to adjust sending rate by changing the inter-packet delay (IPD) at the server side. At every round-trip time, adjustments of sending rate are made in order to achieve responsiveness to sudden changes of buffer availabilities. A series of simulations and the prototype system showed that the proposed methods did not cause buffer overflows and it can maintain smoother rate control and react to bandwidth changes promptly.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2009년도 추계학술발표대회
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• pp.465-466
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• 2009

This paper investigates congestion detection and control strategies for multi-path traffic (CDCM) diss emination in lifetime-constrained wireless sensor networks. CDCM jointly exploits packet arrival rate, succ essful packet delivery rate and current buffer status of a node to measure the congestion level. Our objec tive is to develop adaptive traffic rate update policies that can increase the reliability and the network lif etime. Our simulation results show that the proposed CDCM scheme provides with good performance.

• 한국항행학회논문지
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• 제13권4호
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• pp.523-531
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• 2009

본 논문에서는 패킷 손실 채널로의 전송을 위한 비디오 부호기의 소스/채널 부호화율 통합 제어 방식을 제안한다. 비디오 전송 시스템의 지연 제약에 따른 소스 부호화율 상한을 구하고, 부호화되거나 스킵된 화면에 대한 소스 왜곡과 패킷 손실에 의한 채널 왜곡을 기반으로 일정 부호화 구간에 대한 평균 왜곡을 정의한다. 주어진 채널률과 패킷 손실률에 대해 비디오 부호화 계수와 FEC 블록 크기를 제어계수로 하는 평균 왜곡 최소화 문제를 정의하고, 실시간 해결 알고리즘을 제안한다. 모의실험에서 제안방식은 40% 이하의 패킷 손실률과 30 프레임 이상의 인트라 매크로블럭 갱신률에서 TMN8에 비해 PSNR과 주관적 화질에서 우수한 성능을 보였다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2006년도 심포지엄 논문집 정보 및 제어부문
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• pp.105-107
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• 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.

• 정보처리학회논문지C
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• 제9C권3호
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• pp.375-384
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• 2002

현재의 인터넷은 가변적인 대역폭과 패킷손실 그리고 지연으로 인하여 대화식 응용의 QoS 보장이 어렵다. 특히 최근에 정보의 기반구조로 중요성이 강조되고 있는 VOIP는 패킷손실률과 종점간지연이 클 때 통화품질이 크게 떨어지므로 네트웍 수준에서나 응용 수준에서 에러제어 기법이 요구된다. 인터넷 전화와 같은 대화식 응용을 위한 응용 수준의 에러 제어 기법으로 FEC(Forward Error Correction)가 가장 많이 사용되고 있는데, 이 기법은 주정보와 더불어 부가정보를 전송함으로서 패킷손실을 복구하는 방법으로 네트웍의 상태에 따라 적응적으로 부가정보의 양을 조절한다. 그러나 기존의 알고리즘들은 패킷손실률만을 고려하여 부가정보를 조절하였으며 부가 정보를 증가시킬 때 수반되는 종점간지연을 간과함으로써 통화품질을 떨어뜨리는 단점이 있다. 본 논문에서는 패킷손실률뿐만 아니라 종점간지연을 고려하는 FEC기반 에러제어 기법인 SCCRP (Selecting a Codec Combination using Reward and Penalty)를 제안한다. 실험 결과, SCCRP는 다른 알고리즘들에 비해 복구 후 패킷손실률은 물론 복구 후 종점간지연을 낮게 유지하였다.

• 인터넷정보학회논문지
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• 제18권6호
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• pp.65-73
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• 2017

Congestion occurring at wireless sensor networks(WSNs) causes packet delay and packet drop, which directly affects overall QoS(Quality of Service) parameters of network. Network congestion is critical when important data is to be transmitted through network. Thus, it is significantly important to effectively control the congestion. In this paper, new mechanism to guarantee reliable transmission for the important data is proposed by considering the importance of packet, configuring packet priority and utilizing the settings in routing process. Using this mechanism, network condition can be maintained without congestion in a way of making packet routed through various routes. Additionally, congestion control using packet service time, packet inter-arrival time and buffer utilization enables to reduce packet delay and prevent packet drop. Performance for the proposed mechanism was evaluated by simulation. The simulation results indicate that the proposed mechanism results to reduction of packet delay and produces positive influence in terms of packet loss rate and network lifetime. It implies that the proposed mechanism contributes to maintaining the network condition to be efficient.