• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.75-79
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• 2006

In a VoIP(Voice over IP) system which support the realtime speech service, speech quality is deteriorated by the delay, the jitter, the loss, and the reversed packet order. In this thesis, APBC for receiver site is proposed, which compensate the jitter by the adaptive playout algorithm and conceal the packet loss, and align the packet order. Also, a VoIP application system is implemented, and the performance of APBC is verified on the implemented system by measuring the processing speed and the speech quality. From the result, processing speed is 257$\mu$sec, which is fast enough to deal with packet being received in realtime. Also, the speech quality by MOS(Mean Opinion Score) is improved as 18 percent compared with algorithm of fixed playout delay.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.8
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• pp.3728-3752
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• 2016

Wireless mesh networks (WMNs) have been increasingly applied in private and public networks during the last decade. In a different context, voice over IP (VoIP) has emerged as a new technology for making voice calls around the world over IP networks and is replacing traditional telecommunication systems. The popularity of the two technologies motivated the deployment of VoIP over WMNs. However, VoIP over WMNs suffers from inefficient bandwidth utilization because of two reasons: i) attaching 40-byte RTP/UDP/IP header to a small VoIP payload (e.g., 10 bytes) and ii) 841 μs delay overhead of each packet in WMNs. Among several solutions, VoIP packet multiplexing is the most prominent one. This technique combines several VoIP packets in one header. In this study, we will survey all the VoIP multiplexing methods over WMNs. This study provides a clear understanding of the VoIP bandwidth utilization problem over WMNs, discusses the general approaches in which packet multiplexing methods could be performed, provides a detailed study of present multiplexing techniques, shows the aspects that hinder the VoIP multiplexing methods, discusses the factors affected by VoIP multiplexing schemes, shows the merits and demerits of different multiplexing approaches, provides guidelines for designing a new improved multiplexing technique, and provides directions for future research. This study contributes by providing guidance for designing a suitable and robust method to multiplex VoIP packets over WMNs.

• The Journal of the Korea Contents Association
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• v.7 no.1
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• pp.103-115
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• 2007

The vulnerability issue on IEEE 802.1x wireless LAN has been permits the malicious attack such as Auth/Deauth flooding more serious rather than we expected. Attacker can generate huge volume of malicious traffic as the same methods on existing wired network. The objective of wireless IP Traceback is to determine the real attack sources, as well as the full path taken by the wireless attack packets. Existing IP Traceback methods can be categorized as proactive or reactive tracing. But, these existing schemes did not provide enhanced performance against DoS attack on wireless traffic. In this paper, we propose a 'TTL based advanced Packet Marking' mechanism for wireless IP Packet Traceback with wireless Classification function. Proposed mechanism can detect and control DoS traffic on AP and can generate marked packet for reconstructing on the real path from the non-spoofed wireless attack source, by which we can construct secure wireless network based on AP with enhance traceback performance.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.415-424
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• 2008

In this paper, we compare the QoS performance enhancement schemes according to the network architecture of mobile WiMAX system in order to improve the end-to-end QoS performance and propose QoS parameter mapping method and IP packet scheduling algorithm. To evaluate the end-to-end QoS performance, we implemented an end-to-end simulator of mobile WiMAX system using OPNET. Simulation results show that the proposed QoS parameter mapping scheme reduces the average delay of VoIP packet and improves uplink resource efficiency. And also, when the proposed IP packet scheduling algorithm is applied to the system, the end-to-end packet transmission delay of VoIP service can be reduced by 44-67 percent compared to FIFO and WRR scheduler.

• 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.

• 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.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.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.5-8
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• 2000

The voice over IP(VoIP) technology is currently recognized as the base technology for the next generation telecommunication services. So the VoIP market has been extremely expanding with the opportunity for cheap phone calls. This paper describes the packet voice testing issues and scenarios for the VoIP services. These issues and scenarios are deduced from the testing results through KT VoIP testbed composed of commercial systems.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.209-211
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• 2004

We can categorize mobility two main fields in IP environment. If Mobile IP manages macro mobility, Cellular IP deal with micro mobility. For seamless connection, it is major problem to reduce packet loss in the network layer during handoff. This paper will introduce a scheme which reduces packet loss during micro mobility which use indirect handoff mechanism in Cellular IP, and will verify the efficiency of that scheme by computer simulation.