• Journal of the Korea Society of Computer and Information
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• v.15 no.12
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• pp.75-84
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• 2010

As broadcasting in Mobile Ad hoc NETwork (MANET) is the process that a node sends a packet to all other nodes in the network. it is used for routing protocols such as Ad hoc On demand Distance Vector (AODV) to disseminate control information for establishing the routes. In this paper, we propose Probabilistic Broadcasting mechanism based on Distance Ratio between sender and receive node in MANETs. The proposed approach is based on the combination of probability and distance based approach. A mobile node receiving broadcast packets determines the probability of rebroadcasting considering distance ratio from sender. The distance ratio of a node is calculated by the distance from sender and the length of radio field strength. As a node with high distance ratio is located far away from sender, rebroadcast probability is set to high value. On contrary, the low rebroadcast probability is set for a node with low distance ratio which is close to sender. So it reduces packets transmission caused by the early die-out of rebroadcast packets. Compared with the simple flooding and fixed probabilistic flooding by simulation, our approach shows better performances results. Proposed algorithm can reduce the rebroadcast packet delivery more than 30% without scanting reachability, where as it shows up to 96% reachability compared with flooding.

• The KIPS Transactions:PartB
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• v.9B no.5
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• pp.571-578
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• 2002

In this paper, we propose a video sequence coding scheme called AMV (Auxiliary Motion Vector) to minimize error propagation caused by transmission errors over the Internet. Unlike the conventional coding schemes the AMY coder, for a macroblock in a frame, selects two best matching blocks among several preceding frames. The best matching block, called a primary block, is used for motion compensation of the destination macroblock. The other block, called an auxiliary block, replaces the primary block in case of its loss at the decoder. When a primary block is corrupted or lost during transmission, the decoder can efficiently and simply suppress error propagation to the subsequent frames by replacing the block with an auxiliary block. This scheme has an advantage of reducing both the number and the impact of error propagations. We implemented the proposed coder by modifying H.263 standard coding and evaluated the performance of our proposed scheme in the simulation. The simulation results show that AMV coder is more efficient than the H.263 baseline coder at the high packet loss rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.11A
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• pp.867-875
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• 2003

Current end-to-end congestion control depends only on the information of end points (using three duplicate ACK packets) and generally responds slowly to the network congestion. This mechanism can't avoid TCP global synchronization in which TCP congestion window size is fluctuated during congestion period. Furthermore, if RTT(Round Trip Time) is increased, three duplicate ACK packets are not correct congestion signals because congestion might already disappear and the host may send more packets until it receives three duplicate ACK packets. Recently there are increasing interests in solving end-to-end congestion control using AQM(Active Queue Management) to improve the performance of TCP protocols. AQM is a variation of RED-based congestion control. In this paper, we first evaluate the effectiveness of the current AQM schemes such as RED, CHOKe, ARED, FRED and SRED, over traffic with different rates and over traffic with mixed responsive and non-responsive flows, respectively. In particular, CHOKe mechanism shows greater unfairness, especially when more unresponsive flows exist in a shared link. We then propose a new AQM scheme using CHOKe mechanism, called DQC(Double Queue CHOKe), which uses two FIFO queues before applying CHOKe mechanism to adaptive congestion control. Simulation shows that it works well in protecting congestion-sensitive flows from congestion-causing flows and exhibits better performances than other AQM schemes. Also we use partial state information, proposed in LRURED, to improve our mechanism.

• The KIPS Transactions:PartC
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• v.12C no.4 s.100
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• pp.571-580
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• 2005

In fast long-distance networks, TCP's congestion control algorithm has the problem of utilizing bandwidth effectively. Several window-based congestion control protocols for high-speed and large delay networks have been proposed to solve this problem. These protocols deliberate mainly three properties : scalability, TCP-friendliness, and RTT-fairness. These protocols, however, cannot satisfy above three properties at the same time because of the trade-off among them This paper presents a new window-based congestion control algorithm, called EM (Exponential Increase/ Multiplicative Decrease), that simultaneously supports all four properties including fast convergence, which is another important constraint for fast long-distance networks; it can support scalability by increasing congestion window exponentially proportional to the time elapsed since a packet loss; it can support RTT-fairness and TCP-friendliness by considering RTT in its response function; it can support last fair-share convergence by increasing congestion window inversely proportional to the congestion window just before packet loss. We evaluate the performance of EIMD and other algorithms by extensive computer simulations.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.9 s.351
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• pp.165-174
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• 2006

In this paper, we propose a new contention window control algorithm to increase TCP performance in wireless multi-hop networks. The new contention window control algorithm is suggested to reduce the hidden and exposed terminal problems of wireless multi-hop networks. Most of packet drops in wireless multi-hop networks results from hidden and exposed terminal problems, not from collisions. However, in normal DCF algorithm a failed user increases its contention window exponentially, thus it reduces the success probability of fined nodes. This phenomenon causes burst data transmissions in a particular node that already was successful in packet transmission, because the success probability increases due to short contention window. However, other nodes that fail to transmit packet data until maximum retransmission attempts try to set up new routing path configuration in network layer, which cause TCP performance degradation and restrain seamless data transmission. To solve these problems, the proposed algorithm increases the number of back-of retransmissions to increase the success probability of MAC transmission, and fixes the contention window at a predetermined value. By using ns-2 simulation for the chain and grid topology, we show that the proposed algorithm enhances the TCP performance.

• Journal of the Korea Society of Computer and Information
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• v.19 no.3
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• pp.63-71
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• 2014

We analyze the execution time of Reed-Solomon coding, which is the MAC-layer forward error correction scheme used in CDMA2000 1xEV-DO broadcast services, under different air channel conditions. The results show that the time constraints of MPEG-4 cannot be guaranteed by Reed-Solomon decoding when the packet loss rate (PLR) is high, due to its long computation time on current hardware. To alleviate this problem, we propose three error control schemes. Our static scheme bypasses Reed-Solomon decoding at the mobile node to satisfy the MPEG-4 time constraint when the PLR exceeds a given boundary. Second, dynamic scheme corrects errors in a best-effort manner within the time constraint, instead of giving up altogether when the PLR is high; this achieves a further quality improvement. The third, video-aware dynamic scheme fixes errors in a similar way to the dynamic scheme, but in a priority-driven manner which makes the video appear smoother. Extensive simulation results show the effectiveness of our schemes compared to the original FEC scheme.

• Journal of the Korea Computer Industry Society
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• v.3 no.1
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• pp.95-104
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• 2002

This paper presents some studies on the Internet TCP/IP(Transmission Control Protocol-Internet Protocol) traffic over ATM(Asynchronous Transfer Mode) UBR(Unspecified Bit Rate) and ABR(Available Bit Rate) classes of service. Fuzzy logic prediction has been used to improve the efficiency and fairness of traffic throughput. For TCP/IP over UBR, a novel fuzzy logic based cell dropping scheme is presented. This is referred to as fuzzy logic selective cell drop (FSCD). A key feature of the scheme is its ability to accept or drop a new incoming packet dynamically based on the predicted future buffer condition in the switch. This is achieved by using fuzzy logic prediction for the production of a drop factor. Packet dropping decision is then based on this drop factor and a predefined threshold value. Simulation results show that the proposed scheme significantly improves TCP/IP efficiency and fairness. To study TCP/IP over ABR, we applied the fuzzy logic ABR service buffer management scheme from our previous work to both approximate and exact fair rate computation ER(Explicit cell Rate) switch algorithms. We then compared the performance of the fuzzy logic control with conventional schemes. Simulation results show that on zero TCP packet loss, the fuzzy logic control scheme achieves maximum efficiency and perfect fairness with a smaller buffer size. When mixed with VBR traffic, the fuzzy logic control scheme achieves higher efficiency with lower cell loss.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.2
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• pp.9-17
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• 2012

Because best-effort Internet provides no guarantees on packet delay and loss, transient network congestion may cause negative effects on H.264/SVC streaming. Thus, the congestion control is required to adjust bit rate by dropping enhancement layers of H.264/SVC streams. This paper differentiates the video streams according to different levels of importance and proposes weighted-based congestion control algorithms to use the rate-distortion characteristics of streams. To maximize the weighted sum of PSNR values of all streams on a bandwidth-constrained node, this paper proposes WNS(Weighted Near-Sighted) and WFS(Weighted Far-Sighted) algorithms to control the number of enhancement layers of streams. Through simulation, this paper shows that weighted-based congestion control algorithm can efficiently adapt streams to network conditions and analyzes the characteristics of congestion control algorithms.

• Journal of KIISE:Information Networking
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• v.30 no.6
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• pp.743-754
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• 2003

TFRC(TCP-Friendly Rate Control) is proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with TCP flows[1-3]. However, TFRC has a shortcoming that the fairness and throughput are degraded when the mobile host using TFRC experiences handoffs. This paper proposes a new control mechanism based on TFRC, which deals with the congestion caused by handoffs as well as the losses caused during the handoffs. The simulation results show that our mechanism achieves better throughput and fairness compared to TFRC for repeated handoffs.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.7B
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• pp.901-906
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• 2001

본 논문에서는 이득-투명 SOA 스위치를 이용한 새로운 구조의 광 셀 역압축기를 제안하였다. 제안된 구조는 하드웨어 및 제어의 단순화와 이득-투명 SOA를 이용함으로써 처리용량을 극대화하기 위한 요구조건을 만족시킨다. 제안된 구조의 동작을 실험으로 보이기 위해 4-비트의 소규모 광 압축셀을 역압축시키는 실험을 수행하였으며, 제안구조의 동작원리와 일치하는 결과를 얻을 수 있었다. 이는 시간분할 방식을 포함하는 광 ATM 교환기와 광 패킷망 등에 주요 요소로서 적용될 수 있다.