• The KIPS Transactions:PartC
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• 제9C권3호
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• pp.375-384
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• 2002

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제38C권3호
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• pp.239-246
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• 2013

The IEEE 802.15.4 offers standard about PHY and MAC layer and features low power, low bandwidth, and low speed data communication. Because of this reason, IEEE 802.15.4 is only within a limited range such as sensor detection and home network; nevertheless, the research about transmission multimedia data like voice packet through wireless sensor networks is conducted widely. In this paper, we proposed the group communication system based on the sensor network. TDMA/TDD MAC based on the IEEE 802.15.4 PHY for voice communication on the sensor network is designed by improvement existing peer-to-peer voice communication on the sensor network and hardware is implemented for group communication. To measure the quality of designed system, mean opinion score (MOS) is obtained from the experiment and verified by using sine wave method. As a result of an experiment, we expect that a many cases of application solution can be developed using presented system.

• 한국정보통신설비학회:학술대회논문집
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• 한국정보통신설비학회 2009년도 정보통신설비 학술대회
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• pp.99-103
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• 2009

Recently, as wired and wireless communication services have rapidly developed and multimodal mobile devices which have various characteristics have widely spread, the need for new convergence services increases. The growing population of VoIP technologies and the high communication expense yield that the market of IP based telephony such as WiFi phone and IP phone is substituted for one of the conventional PSTN telephony. With the help of this trend, the wireline network operators desire to find a market in mobile networks. Therefore, they focus on Fixed Mobile Convergence (FMC) service as one of the key factors to accomplish this goal. FMC services are able to provide the mobility of voice services between circuit switched and packet switched networks. IP Multimedia Subsystem (IMS) based Voice Call Continuity (VCC) is one of the schemes to embody FMC services. As Application Server (AS) which has this VCC function provides seamless handover of services between heterogeneous networks, FMC subscribers can communicate seamlessly with others m WiFi domain and COMA domain using WiFi-COMA dual phone. Most of enterprises have already introduced IP network infrastructure and IP-PBX (Private Branch eXchange) for telephony. However, the problems of high communication cost and work inefficiency due to frequent outside jobs or business trips have remained. In order to solve these problems, demands for enterprise FMC services increase. In this paper, we introduce a new IP-PBX based VCC model that can provide seamless handover of voice services between WiFi and COMA networks for enterprise users and we investigate some interworking and security issues between Soft Switch (SSW) and IMS, or between IMSs. In addition, we introduce a new service that can provide the continuity of voice sessions as well as video sessions using Multimedia Session Continuity (MMSC) technology which has evolved from VCC. This service is expected to be one of the next-generation personalized services based on user's context.

• Journal of the Institute of Electronics Engineers of Korea SD
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• 제42권2호
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• pp.89-96
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• 2005

In this paper, we propose a customer premise platform for processing multimedia data service on the ATM network. The proposed platform has a specific AAL2 processor that includes AAL2 protocol and scheduler algorithm so as to off-load large potion of burden from host processor and make it easy to process multimedia data from the ATM network in real time compared with conventional platform in which AAL/ATM tasks are processed by software. The ATS scheduler that is implemented based on 2-level time slot ring provides a simple and efficient method for scheduling data of VBR-rt, UBR and CBR traffics. TMS320C5402 DSP is used to process voice-related tasks such as voice compression and voice packet manupulation and AAL2 processor is implemented on $0.35\;{\mu}m$ process line. We implemented the customer premise equipment for VoDSL service and tested the proposed platform on a test bed network. The experimental results show that the proposed equipment has the call success rate of $97\%$ at least and provides voice service of toll-qualify.

• Journal of the Institute of Electronics Engineers of Korea TC
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• 제42권12호
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• pp.127-134
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• 2005

Lost packet robustness is an most important quality measure for voice over IP networks(VoIP). Recovery of the lost packet from the received information is crucial to realize this robustness. So, this paper proposes the lost packet recovery method from the received information for real-time communication for CELP coder. The proposed BS-PLC (Both Side Packet Loss Concealment) based WSOLA(Waveform Shift OverLab Add) allow the lost packet to be recovered from both the 'previous' and 'next' good packet as the LP parameter and the excitation signal are respectively recovered. The burst of packet loss is modeled by Gilbert model. The proposed scheme is applied to G.729 most used in VoIP and is evaluated through the SNR(signal to noise) and the MOS(Mean Opinion Score) test. As a simulation result, The proposed scheme provide 0.3 higher in Mean Opinion Score and 2 dB higher in terms of SNR than an error concealment procedure in the decoder of G.729 at $20\%$ average packet loss rate.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제30권9A호
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• pp.737-748
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• 2005

Two flexible admission control schemes for integrated voice and stream-type data services are proposed in DS-CDMA systems. Most Previous studies on admission control have focused on integration of short, bursty Packet-type data services and conventional voice services. However, stream-type data services with a relatively long service holding time are expected to be a considerable portion of data traffic in future generation cellular systems. Scheme I is a basic scheme that accommodates both voice and data services with full bandwidth. However, voice services are given priority over data services using the duration difference between the holding times for these services. Scheme ll uses a different method to efficiently give priority to voice services over stream-type data services. An additional interference margin for voice services is provided by suppressing interference from stream-type data services according to voice access requests and a varying interference status. Performance of the two schemes is evaluated by developing Markovian models. Numerical results show that the voice capacity is highly sensitive to the service holding time of data services while the performance measures of data services are not highly sensitive. Scheme H is a significant improvement over Scheme I for accommodating voice and stream-type data services

• Proceedings of the IEEK Conference
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• 대한전자공학회 1998년도 추계종합학술대회 논문집
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• pp.55-58
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• 1998

In this paper, we describe the design of the ATM adapter circuit in the BSC for IMT-2000 Network. This ATM adapter circuit can convert received ATM cell into TDM data in the BSC and vice versa. In the ATM adapter, we implemented both AAL1 and AAL5 functions to provide constant bit rate voice data and variable bit rate packet data servives, simultaneously.

• Proceedings of the Korean Information Science Society Conference
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• 한국정보과학회 2004년도 가을 학술발표논문집 Vol.31 No.2 (3)
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• pp.331-333
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• 2004

본 논문에서는 SIP(Session initiation Protocol) 기반의 VoIP(Voice over Internet Protocol) 시스템에 효율적인 call-control을 위해 필요한 헤더와 파라미터를 추가한 Extension SIP를 제안하였다. 또한 이 제시된 Extension SIP에 따르는 SIP Proxy Server와 User Agent(User Agent Client, User Agent Server)를 리눅스 시스템에서 C언어를 통해 구현하였고, 이 구현된 Extension SIP 시스템을 통해 기존의 SIP 시스템과 cail-control을 위한 packet traffic을 비교.분석 함으로써 제안한 Extension SIP의 효율성 을 증명하였다.

• Information and Communications Magazine
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• 제19권7호
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• pp.40-55
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• 2002

The advent of Third Generation (3G) communication systems, with their ability to process real-time multimedia applications and their large bandwidths, will greatly enhance mobile Internet access. Not only does Wideband Code Division Multiple Access (WCDMA) and cdma2000 radio technology offer an advantageous density for voice in terms of spectral efficiency, it also supports higher rates and offers differentiated levels of duality of Service (QoS) for data applications. The early introduction of packet and multimedia technologies will be a key element in realizing a quick and successful return on the operator's investments in Universal Mobile Telecommunications System (UMTS) and cdma2000.

• Journal of the Institute of Electronics Engineers of Korea TC
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• 제41권10호
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• pp.1-9
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• 2004

User-Perceived application-level performance is a key to the adoption and success of CDMA 2000. To predict this performance in advance, a detailed end-to-end simulation model of a CDMA network was built to include application traffic characteristics, network architecture, network element details, and protocol features. We assess the user application performance when a Radio Access Network (RAN) and a Core Network (CN) adopt different transport architectures such as ATM and If. For voice Performance, we found that the vocoder bypass scenario shows 8% performance improvement over the others. For data packet performance, we found that HTTP v.1.1 shows better performance than that of HTTP v.1.0 due to the pipelining and TCP persistent connection. We also found that If transport technology is better solution for higher FER environment since the IP packet overhead is smaller than that of ATM for web browsing data traffic, while it shows opposite effect to small size voice packet in RAN architecture. Though simulation results we showed that the 3G-lX EV system gives much better packet delay performance than 3G-lX RTT, the main conclusion is that end-to-end application-level performance is affected by various elements and layers of the network and thus it must be considered in all phases of the technology evolution process.