• Journal of IKEEE
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• v.2 no.1 s.2
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• pp.68-75
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• 1998

The Asynchronous Transfer Mode(ATM) networks are being adopted as backbones over various parts of Internet. Also, TCP is one of the most widespread transport protocols, nowadays. It can be used with ATM. But, TCP shows poor end-to-end performance on ATM networks. Effective throughput of TCP over ATM can be quite low when cells are dropped at the congested ATM switch. The low throughput is due to wasted bandwidth as congested link transmits cells from corrupted packets. This paper examines the behavior of TCP over ATM-UBR using EPD switch in a broadband environment. With a threshold close to the buffer size, the buffer can be used more efficiently, but more drops and retransmission occurs. If the threshold is much less than buffer size, efficient is not good, but few drops can happen. Therefore, decision of the threshold is important.

• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.401-410
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• 2007

Multimedia streams transmitted through the Internet have a strict playback delay time. Multimedia data arriving aster the playback time cannot be played in the receiver and are discarded. Thus, this paper proposed a protocol, in which the multimedia stream server determines whether data can be played in the receiver before sending the data. The proposed model has a PSCTP sub-layer on top of existing PR-SCTP and decides whether to send data messages, which have come from the multimedia applicationserver, and which PR-SCTP stream the data will be sent to. In addition, the proposed model uses the differentiated retransmission function of PR-SCTP. We evaluated the performance of SCTP, PR-SCTP and PSCTP using NS2 simulator. According to the results of the evaluation, the PSCTP protocol decreased the volume of transmission and increased the video decodable ratio compared to other protocols.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.916-919
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• 2003

Reliable multicast data transmission in a 1:N environment needs more sophisticated error control mechanism than that of in 1:1 environment due to ACK implosion and duplicated retransmission. Although there have been many related research on error control in reliable multicast, real implemented protocols are rare. As one of the reliable multicast transport protocols, ECTP is selected as an international standard reliable multicast protocol by ITU-T and ISO and implemented on RedHat 7.2 machine by us. In this paper, we evaluate the performance of the error control mechanism in the respect of throughput and generated control packet numbers with a real implementation code. From the results, it is concluded that the suitable values of error control parameters can be obtained from the local group size and network environments.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.10a
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• pp.681-684
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• 2008

Bluetooth baseband performs FEC (forward error check) at the interface of transmitter and receiver modem. Well-designed FEC means directly the efficiency of retransmission of the data payload therefore design optimization is very important. In this paper, we designed a optimal 1/3, 2/3 type of FEC. 1/3 FEC. which performs 3 times customary repetition was designed for packet header, and 2/3 FEC was designed for data packets with (15, 10) reduced hamming code. The proposed hardware FEC block was described and verified using Verilog HDL and later to be automatically synthesized. The synthesized FEC block operated at 40Mhz normal clock speed of the target baseband microcontroller.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.13 no.6
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• pp.535-544
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• 1988

In this paper, the hybrid ARQ scheme, which is incorporated the selective-repeat ARQ system with the finite receiver buffer and the single-error-correcting and double-error-detecting(63.56) cyclic Hamming code system, has been investigated. As a result of simulation, the proposed hybrid ARQ scheme shows that that throughput efficiencies are improved by one error correction, and that the reversed codewords due to retransmission are delivered to the user in order by means of detecting two errors. The hybrid ARQ scheme significantly outperforms the FEC or the ideal selective-repeat ARQ system in the respect of throughput and reliability, especially when the channel error rate is approximately in the range from $10^{-2}$~$10^{-3}$.

• Journal of Communications and Networks
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• v.6 no.3
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• pp.235-244
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• 2004

There have been a lot of approaches to evaluate and predict transmission control protocol (TCP) performance in a numerical way. Especially, under the recent advance in wireless transmission technology, the issue of TCP performance over wireless links has come to surface. It is because TCP responds to all packet losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. By several previous works, although it has been already proved that overall TCP performance is largely dependent on its loss recovery performance, there have been few works to try to analyze TCP loss recovery performance with thoroughness. In this paper, therefore, we focus on analyzing TCP's loss recovery performance and have developed a simple model that facilitates to capture the TCP sender's behaviors during loss recovery period. Based on the developed model, we can derive the conditions that packet losses may be recovered without retransmission timeout (RTO). Especially, we have found that TCP Reno can retransmit three packet losses by fast retransmits in a specific situation. In addition, we have proved that successive three packet losses and more than four packet losses in a window always invoke RTO easily, which is not considered or approximated in the previous works. Through probabilistic works with the conditions derived, the loss recovery performance of TCP Reno can be quantified in terms of the number of packet losses in a window.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1609-1617
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• 2013

TFRC(TCP-Friendly Rate Control) has a performance degradation in wireless networks because it performs congestion control by judging all the losses occurred in wireless networks as a congestion indicator. It is also degraded by the increased Round Trip Time(RTT) due to packet retransmission and contention overhead in the link layer. In this paper, we propose an adaptive rate control scheme based on cross-layer to improve the quality of streaming services in the wireless networks. It provides new RTT estimation and loss discrimination methods to improve transmission rate of TFRC. The simulation results show that the proposed scheme can improve the performance of TFRC.

• Journal of KIISE:Information Networking
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• v.31 no.2
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• pp.224-232
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• 2004

To guarantee the reliability in multicast transmission, researches providing reliability through hierarchical control tree which is independent on data channel tree are known to provide high scalability. However, the logical control tree in transport layer constructed without topology information of the corresponding network layer tree may inefficiently use the network resources because the logical control tree is not closely related to the tree topology of the network layer. A router assisted control tree mechanism presented in this paper would improve the efficiency of the link as well as it would remove the replicated data. In addition, it requires to a router a small change which examines the message type of the control tree.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4A
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• pp.330-335
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• 2007

This paper proposes a multichannel random access channel allocation scheme for multihop cellular networks to guarantee the stable throughput of a random access. The fundamental contribution is a mathematical formula for an optimal partition ratio of shared random access channels between a base station and a relay station. In addition, the proposed scheme controls the retransmission probability of random access packets under heavy load condition. Simulation results show that the proposed scheme can guarantee the required random access channel utilization and packet transmission delay even if the a random access packet arrival rate is higher than 0.1.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.4
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• pp.279-290
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• 2013

In this paper, we propose a transmission scheme of MAC protocol that enables secure voice communications by exploiting the wide spectrum and low signal strength characteristics of the ultra wide band technology. In addition, it also supports high level of terminal mobility by deploying mobile ad hoc network schemes. While most of existing UWB MAC protocols are operated as a synchronous mode, the proposed scheme operates in an asynchronous mode for supporting high mobility and sends voice packets without RTS/CTS control packets for efficient voice traffic transmission without retransmission. With simulation program, we prove that the proposed scheme satisfies the required voice quality and packet delivery time.