• The Journal of the Acoustical Society of Korea
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• v.32 no.2
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• pp.157-166
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• 2013

This paper proposes a non-intrusive speech quality estimation method considering the effects of packet loss to perceptual quality. Packet loss is a major reason of quality degradation in a packet based speech communications network, whose effects are different according to the input speech characteristics or the performance of the embedded packet loss concealment (PLC) algorithm. For the quality estimation system that involves packet loss effects, we first observe the packet loss of G.729 codec which is one of narrowband codec in VoIP system. In order to quantify the lost packet affects, we design a classification algorithm only using speech parameters of G.729 decoder. Then, the degradation values of each class are iteratively selected that maximizes the correlation with the degradation PESQ-LQ scores, and total quality degradation is modeled by the weighted sum. From analyzing the correlation measures, we obtained correlation values of 0.8950 for the intrusive model and 0.8911 for the non-intrusive method.

• ETRI Journal
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• v.38 no.6
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• pp.1064-1073
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• 2016

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.9B
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• pp.1272-1280
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• 2010

WFQ (Weighted Fair Queuing) provides not only fairness among traffic flows in using bandwidth but also guarantees the Quality of Service (QoS) that individual flow requires, which is why it has been applied to the resource reservation protocol (RSVP)-capable router. The RSVP allocates an enough resource to satisfy both the rate and end-to-end delay requirements of the flow in condition of no packet loss, and the WFQ guarantees those QoS requirements with the allocated resource. In a practice, however, most QoS-guaranteed services, specially the Voice of IP, allow a few percent of packet loss, so it is strongly desired that the RSVP and WFQ make the best use of this allowable packet loss. This paper enhances the WFQ to allow packet loss and investigates its performance. The performance evaluation showed that allowing the packet loss of 0.4% can improve the flow admission capability by around 40 percent.

• 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.

• ETRI Journal
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• v.25 no.5
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• pp.356-368
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• 2003

This paper analyzes the characteristics of packet losses in mobile computing environments based on the Gilbert model and then describes a mechanism that can recover the lost audio packets using redundant data. Using information periodically reported by a receiver, the sender dynamically adjusts the amount and offset values of redundant data with the constraint of minimizing the bandwidth consumption of wireless links. Since mobile computing environments can be often characterized by frequent and consecutive packet losses, loss recovery mechanism need to deal efficiently with both random and consecutive packet losses. To achieve this, the suggested mechanism uses relatively large, discontinuous exponential offset values. That gives the same effect as using both the sequential and interleaving redundant information. To verify the effectiveness of the mechanism, we extended and implemented RTP/RTCP and applications. The experimental results show that our mechanism, with an exponential offset, achieves a remarkably low complete packet loss rate and adapts dynamically to the fluctuation of the packet loss pattern in mobile computing environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.10
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• pp.1508-1517
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• 1993

In this paper, we propose the modeling of a wireless communication system using CSMA protocol, present analytical evalution and simulation as a function of arrival rate and mean END-to-END delay in the distributed packet radio network. Asynchronous 1-persistent CSMA protocol is used in wireless communication system with half duplex. We assume that all terminals are to be in the close range of each other, suitably located in the local area. The traffic presented to a common channel is assumed to be poisson distribution. Analytical model is based on a M/D/1 with breakdown. In conclusion for wireless network model proposed in this paper is suitable for packet arrival rate of 2 packet/sec with mean packet delay time less than 2 times the packet transmission time.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.367-376
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• 2005

The smooth handoff supported by the route optimization extension to the mobile IP standard protocol should support a packet buffering mechanism at the base station (BS), in order to reduce the degradation in TCP performance caused by packet losses within mobile network environments. The purpose of packet buffering at the BS is to recover the packets dropped during intersubnetwork handoff by forwarding the packets buffered at the previous BS to the new BS. However, when the mobile host moves to a congested BS within a new foreign subnetwork, the buffered packets forwarded by the previous BS are likely to be dropped. This subsequently causes global synchronization to occur, resulting in the degradation of the wireless link in the congested BS, due to the increased congestion caused by the forwarded burst packets. Thus, in this paper, we propose an implicit priority forwarding (IPF) packet buffering scheme as a solution to this problem within mobile IP based networks. In the proposed IPF method, the previous BS implicitly marks the priority packets being used for inter-subnetwork handoff. Moreover, the proposed modified random early detection (M-RED) buffer at the new congested BS guarantees some degree of reliability to the priority packets. The simulation results show that the proposed IPF packet buffering scheme increases the wireless link utilization and, thus, it enhances the TCP throughput performance in the context of various intersubnetwork handoff cases.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.3B
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• pp.213-224
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• 2009

Most of existing routing methods in wireless sensor networks to counter the local eavesdropping-based packet-tracing deal with a single asset and suffer from the packet-delivery latency as they prefer to take a separate path of many hops for each packet being sent. Recently, the author proposed a routing method, GSLP-w(GPSR-based Source-Location Privacy with crew size w), that enhances location privacy of the packet-originating node(i.e., active source) in the presence of multiple assets, yet taking a path of not too long. In this paper, we present a refined routing(i.e., next-hop selection) procedure of it and empirically study privacy strength and delivery latency with varying the crew size w(i.e., the number of packets being sent per path). It turns out that GSLP-w offers the best privacy strength when the number of packets being sent per path is randomly chosen from the range [$1,h_{s-b}/4$] and that further improvements on the privacy are achieved by increasing the random walk length TTLrw or the probability prw that goes into random walk(where, $h_{s-b}$ is the number of hops of the shortest path between packet-originating node s and sink b).

• The Journal of the Acoustical Society of Korea
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• v.16 no.6
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• pp.48-53
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• 1997

The multicasting of MCS(Multicast Server) requires a effective traffic control scheme to prevent buffer overflow on ATM subnetworks. This paper considers MCS multicasting to TCP packets, and propose EPD + SPD scheme(Early Packet Discard-same Source Packet Discard) using common buffer. When the threshold of output buffer is reached, MCS drops an entire packet prior to buffer overflow, so that corrupted packets will not be transmitted by the server. And SPD scheme show that the EPD + SPD results in higher TCP throughput than that of tail drop and EPD + DFF.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.11C
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• pp.1171-1177
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• 2002

In this paper, we proposed a high speed IP(Internet Protocol) packet forwarding engine of ATM(Asynchronous Transfer Mode) based label edge routers for POS(Packet over SONET) interface. The forwarding engine uses TCAM(Ternary Content Addressable Memory) for high performance lookup processing of the packet received from POS interface. We have accomplished high speed IP packet forwarding in hardware by implementing the functions of high speed IP header Processing and lookup control into FPGA(Field Programmable Gate Array). The proposed forwarding engine has the functions of label edge routers as the lookup controller supports MPLS(Multiprotocol Label Switching) packet processing functionality.