• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.4C
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• pp.343-355
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• 2003

Fair bandwidth allocations at routers protect adaptive flows from non-adaptive ones and may simplify end-to end congestion control. However, traditional fair bandwidth allocation mechanisms, like Weighted Fair Queueing and Flow Random Early Drop, maintain state, manage buffera and perform packet scheduling on a per-flow basis. These mechanisms are more complex and less scalable than simple FIFO queueing when they are used in the interi or of a high-speed network. Recently, to overcome the implementation complexity problem and address the scalability and robustness, several fair bandwidth allocation mechanisms without per-flow state in the interior routers are proposed. Core-Stateless Fair Queueing and Rainbow Fair Queuing are approximates fair queueing in the core-stateless networks. In this paper, we proposed simple Layered Fair Queueing (SLFQ), another core-stateless mechanism to approximate fair bandwidth allocation without per-flow state. SLFQ use simple layered scheme for packet labeling and has simpler packet dropping algorithm than other core-stateless fair bandwidth allocation mechanisms. We presente simulations and evaluated the performance of SLFQ in comparison to other schemes. We also discussed other are as to which SLFQ is applicable.

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.649-655
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• 2003

Network-based computing such as cluster computing requires a reliable high-speed transport protocol. TCP is a representative reliable transport protocol on the Internet, which implements many mechanisms, such as flow control, congestion control, and retransmission, for reliable packet delivery. This paper, however, finds out that Myrinet does not incur any packet losses caused by network congestion. In addition, we ascertain that Myrinet supports reliable and ordered packet delivery. Consequently, most of reliable routines implemented in TCP produce unnecessarily additional overheads on Myrinet. In this paper, we show that we can attain the reliability only by flow control on Myrinet and propose a new reliable protocol based on UDP named RUM (Reliable UDP on Myrinet) that performs a flow control. As a result, RUM achieves a higher throughput by 45% than TCP and shows a similar one-way latency to UDP.

• The Journal of the Korea Contents Association
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• v.10 no.8
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• pp.59-65
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• 2010

This paper proposes an implementation of a hardware module for transmitting MPEG-2 TS data over the internet protocol (IP)-based network. This implementation consists of two modules; one is an encapsulation module which bridges between n TS packets, where $1\;{\leq}\;n\;{\leq}\;7$, and an IP packets, the other is a packet conversion module which extracts an DSM-CC PS packet from consecutive TS packets and then reconstructing an IP packet. So, these IP packets are carried over 150 megabits per second. Although overall work flow of the proposed DeMUX is based on the reference design of ALTERA, the DeMUX is enhanced by modifying it and performs more functions by adding a packet conversion module. The DeMUX is described by Verilog-HDL (hardware description language) and shows the faithful functionality and throughput through the simulation.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.32A no.2
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• pp.11-18
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• 1995

Error detection in ARQ(Automatic Repeat Request) protocols is very important in wireless data communication systems. The throughput efficiency of ARQ protocols can be improved by dynamically adapting the protocol packet length so that it approaches the optimum value for throuhput efficiency. In this paper, a simple and novel adaptive packet length allocation method is proposed which transmits the packets with variable length by dyanmically estimating the channel codition. The simulation results show that the average of throughput is improved by 315.4% in the stop-and-wait protocol, 41.4% in the go-back-N protocol and 155.9% in the selective repeat protocol respectively. And the throughput performances of adaptive packet length allocation method approximately approach the theoritically optimal throughput performances.

• IE interfaces
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• v.4 no.1
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• pp.63-69
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• 1991

In this paper, the Go-Back-N ARQ protocol with decoding in communication network is considered. The time delay and throughput are respectively analyzed as a function of window size and decoding time out. Packets arrive continuously at the decoder, and are stored in a buffer if the decoder is busy upon its arrival. The decoder devotes no more than a time-out period of predetermined length to the decoding of any single packet. If packet decoding is completed within that period, the packet leaves the system. Otherwise, it is retransimitted and its decoding starts anew. The time delay and throughput are obtained using recursive formula and difference equation. An appropriate time out and window size that satisfies the grade of service can be determined.

• Journal of Internet Computing and Services
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• v.3 no.6
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• pp.43-51
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• 2002

In this paper, we study the new packet dropping scheme using an active queue management algorithm. Active queue management mechanisms differ from the traditional drop tail mechanism in that in a drop tail queue packets are dropped when the buffer overflows, while in active queue management mechanisms, packets may be dropped early before congestion occurs, However, it still incurs high packet loss ratio when the buffer size is not large enough, By detecting congestion and notifying only a randomly selected fraction of connection, RED causes to the global synchronization and fairness problem. And also, it is the biggest problem that the network traffic characteristics need to be known in order to find the optimum average queue length, We propose a new efficient packet dropping method based on the active queue management for congestion control. The proposed scheme uses the per-flow rate and fair share rate estimates. To this end, we present the estimation algorithm to compute the flow arrival rate and the link fair rate, We shows the proposed method improves the network performance because the traffic generated can not cause rapid fluctuations in queue lengths which result in packet loss

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.11
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• pp.1177-1184
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• 2014

An wired ship area network has functionality of remote control and autonomous management of various sensors and instruments embedded or boarded in a ship. For such environment, the DiffServ (Differentiated Services) realizes that the high-speed real-time flow with the higher priority has the guaranteed minimum data rate and is delivered faster. As a result of this DiffServ effect, the intelligent Ship Area Networks can be implemented. In this paper, an packet drop technique is proposed to outperform the previous RIO (RED In and Out) drop mechanism for DiffServ in ship area networks. the proposed packet drop technique does not manage the individual flows and divides them into several flow groups according to a criterion. And it guarantees the fairness between individual flows in the same QoS class through the group-based control. In simulation results of the proposed packet drop technique, the link utilization decreases than RIO. But it guarantees more data rates to DiffServ flows passing multiple bottleneck links.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8A
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• pp.787-793
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• 2008

3GPP evolved packet system (EPS) is an all-IP based system that supports various access networks such LTE, HSPA/HSPA+, and non-3GPP networks. Recently, the support of IP flows with packet level QoS profiles was added to the requirements of the EPS. This paper proposes an adaptive modulation and coding (AMC) scheme that supports the QoS of such IP flows in the 3G LTE access network of the EPS. Defining the retransmission as a critical factor for QoS, the proposed scheme applies different maximum packet error probability $P_{max}$ to each packet when selecting the AMC transmission mode. In determining $P_{max}$, the QoS constraints and NACK-to-ACK error as well as channel condition are considered, balancing two objectives: the satisfaction of the QoS and the maximization of spectral efficiency. The simulation results show that it is able to reduce both delay violation and status report by 10%, while improving the throughput 10% in comparison with an existing scheme.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.91-100
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• 2004

This paper presents a receiver-driven TCP flow control mechanism, which is adaptive to the wireless condition, for memory constrained mobile receiver. A receiver-driven TCP flow control is, in general, achieved by adjusting the size of advertised window at the receiver. The proposed method constantly measures at the receiver both the available wireless bandwidth and the packet round-trip time. Depending on the measured values, the receiver adjusts appropriately the size of advertised window. Constrained by the adjusted window which reflects the current state of the wireless network, the sender achieves an improved TCP throughput as well as the reduced round-trip packet delay. Its implementation only affects the protocol stack at the receiver and hence neither the sender nor the router are required to be modified. The mechanism has been implemented in real environments. The experimental results show that in CDMA2000 1x networks the TCP throughput of the proposed method has improved about 5 times over the conventional method when the receiver's buffer size is limited to 2896 bytes. Also, with 64Kbytes of buffer site, the packet round-trip time of the proposed method has been reduced in half, compared the case with the conventional method.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.1
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• pp.79-95
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• 2016

In order to improve the Quality-of-Service (QoS) of latency sensitive applications in next-generation cellular networks, multi-path is adopted to transmit packet stream in real-time to achieve high-quality video transmission in heterogeneous wireless networks. However, multi-path also introduces two important challenges: out-of-order issue and reordering delay. In this paper, we propose a new architecture based on Software Defined Network (SDN) for flow aggregation and flow splitting, and then design a Multiple Access Radio Scheduling (MARS) scheme based on relative Round-Trip Time (RTT) measurement. The QoS metrics including end-to-end delay, throughput and the packet out-of-order problem at the receiver have been investigated using the extensive simulation experiments. The performance results show that this SDN architecture coupled with the proposed MARS scheme can reduce the end-to-end delay and the reordering delay time caused by packet out-of-order as well as achieve a better throughput than the existing SMOS and Round-Robin algorithms.