• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.7B
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• pp.528-535
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• 2012

This paper proposes silence drop first(SDF) active buffer management algorithm to increase the voice capacity when silence suppression is used. This algorithm finds and drops silence packet rather than voice packet in the queue for resolving buffer overflow of queue. Simulations with voice codec of G.729A and G.711 are performed. By using proposed SDF algorithm, the voice capacity is increased by 84.21% with G.729A and 38.46% with G.711. Further more, SDF algorithm reduces the required link capacity and loosens the silence packet inter-arrival time limit to provide target voice quality compared with that of conventional algorithms.

• Journal of Internet Computing and Services
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• v.15 no.2
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• pp.9-18
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• 2014

The sensors attached on/in a person are moved since human body frequency changes their activity, therefore in wireless body area networks, nodal mobility and non-line-of-sight condition will impact on performance of networks such as energy efficiency and reliable communication. We then proposed schemes which study on forwarding decisions against frequent change of topology and channel conditions to increase reliable connections and improve energy efficiency. In this work, we control the size of packets, forwarding rate based on ratio of input links and output links at each node. We also robust the network topology by extending the peer to peer IEEE 802.15.4-based. The adaptive topology from chain-based to grid-based can optimal our schemes. The simulation shows that these approaches are not only extending network lifetime to 48.2 percent but also increase around 6.08 percent the packet delivery ratio. The "hot spots" problem is also resolved with this approach.

• Journal of Information Processing Systems
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• v.13 no.5
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• pp.1229-1242
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• 2017

In this paper, an interference aware distributed multi-channel MAC (IDMMAC) protocol is proposed for wireless sensor and actor networks (WSANs). The WSAN consists of a huge number of sensors and ample amount of actors. Hence, in the IDMMAC protocol a lightweight channel selection mechanism is proposed to enhance the sensor's lifetime. The IDMMAC protocol divides the beacon interval into two phases (i.e., the ad-hoc traffic indication message (ATIM) window phase and data transmission phase). When a sensor wants to transmit event information to the actor, it negotiates the maximum packet reception ratio (PRR) and the capacity channel in the ATIM window with its 1-hop sensors. The channel negotiation takes place via a control channel. To improve the packet delivery ratio of the IDMMAC protocol, each actor selects a backup cluster head (BCH) from its cluster members. The BCH is elected based on its residual energy and node degree. The BCH selection phase takes place whenever an actor wants to perform actions in the event area or it leaves the cluster to help a neighbor actor. Furthermore, an interference and throughput aware multi-channel MAC protocol is also proposed for actor-actor coordination. An actor selects a minimum interference and maximum throughput channel among the available channels to communicate with the destination actor. The performance of the proposed IDMMAC protocol is analyzed using standard network parameters, such as packet delivery ratio, end-to-end delay, and energy dissipation, in the network. The obtained simulation results indicate that the IDMMAC protocol performs well compared to the existing MAC protocols.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.8
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• pp.1384-1390
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• 2008

The TCP protocol is originally designed for wired network, however it performs very poor in wireless network due to different nature of wireless network from wired networks. In terms of TCP performance improvement in wireless Ad-hoc network, many researches show that small congestion window size of TCP connection can improve TCP performance. We propose a new TCP algorithm to improve TCP performance in wireless Ad-hoc network. The basic idea of our approach is that TCP receiver estimates the optimum window size and then sets congestion window limit of TCP sender to an optimum value by using the advertised window field in TCP ACK packet. From extensive computer simulation, the proposed algorithm shows superior performance than traditional TCP protocols in terms of packet delivery ratio and packet loss.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.4
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• pp.581-587
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• 2021

Real-time video streaming scheme for multimedia content delivery and remote conference services is one of technologies that are significantly sensitive to data transmission delay. Recently, because of COVID-19, real-time video streaming contents for the services are significantly increased such as personal broadcasting and remote school class. In order to support the services, there is a growing emphasis on low transmission delay and secure content delivery, respectively. Therefore, our research proposed a packet aggregation algorithm to reduce the transmission delay of selectively encrypted video transmission for real-time video streaming services. Through the application of the proposed algorithm, the selectively encrypted video framework can control the amount of MPEG-2 TS packets for low latency transmission with a consideration of packet priorities. Evaluation results on testbed show that the application of the proposed algorithm to the video framework can reduce approximately 11% of the transmission delay for high and low resolution video.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.14 no.4
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• pp.247-256
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• 2021

In this paper, we propose a packet loss concealment (PLC) algorithm using pitch harmonic motion prediction and adaptive signal amplitude prediction and. The spectral motion prediction method divides the spectral motion of the previous usable frame into predetermined sub-bands to predict and restore the motion of the lost signal. In the proposed algorithm, the speech signal is classified into voiced and unvoiced sounds. In the case of voiced sounds, it is further divided into pitch harmonics using the pitch frequency to predict and restore the pitch harmonic motion of the lost frame, and for the unvoiced sound, the lost frame is restored using the spectral motion prediction method. When the continuous loss of speech frames occurs, a method of adjusting the gain using the least mean square (LMS) predictor is proposed. The performance of the proposed algorithm was evaluated through the objective evaluation method, PESQ (Perceptual Evaluation of Speech Quality) and was showed MOS 0.1 improvement over the conventional method.

• Journal of KIISE:Information Networking
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• v.36 no.1
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• pp.34-42
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• 2009

In this paper, we propose a practical hybrid transport error control scheme to provide SVC video streaming service over error-prone IP networks. Many error control mechanisms for various video coding standards have been proposed in the literature. However, there is little research result which can be practically applicable to the multilayered coding structure of SVC(the scalable extension of H.264/AVC). We present a new hybrid transport error control scheme that efficiently combines layered Forward Error Correction(FEC) and Automatic Repeat Request(ARQ) for better packet-loss resilience. In the proposed hybrid error control, we adopt ACK-based ARQ instead of NACK-based ARQ to maximize throughput which is the amount of effective data packets delivered over a physical link per time unit. In order to prove the effectiveness of the proposed hybrid error control scheme, we adopt NIST-Net network emulator which is a general-purpose tool for emulating performance dynamics in IP networks. It is shown by simulations over the NIST-Net that the proposed hybrid error control scheme shows improved packet-loss resilience even with much less number of overhead packets compared to various conventional error control schemes.

• Journal of information and communication convergence engineering
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• v.5 no.2
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• pp.88-92
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• 2007

In this paper, an SS-RRA protocol that is based on Code Division Multiple Access is proposed and analyzed under the integrated voice and data traffic load. The backward logical channels consist of slotted time division frames with multiple spreading codes per slot. The protocol uses a reservation mechanism for the voice traffic, and a random access scheme for the data traffic. A discrete-time, discrete-state Markov chain is used to evaluate the performance. The numerical results show that the performance can be significantly improved by a few distinct spreading codes.

• Proceedings of the IEEK Conference
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• 2003.07b
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• pp.843-846
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• 2003

In this paper, we have analyzed algorithm of physical layer, data link layer and MAC layer of Out-Of-Band (OOB) specified in the OpenCabl $e^{TM}$SCTE-55-2 2002$^{[3]}$ and designed architecture of the OOB processor. The OOB processor performs fundamental multiple access control for the OOB channel and extracts session key information from a EMM packet for descrambling MPEG-2 streams. In this paper, we have implemented a prototype board for the DVS167 OOB processor and verified it.t.