• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.928-938
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• 2008

Though services requiring Quality-of-Service (QoS) guarantees such as Voice over Internet Protocol (VoIP) have been widely deployed on the internet, most of internet backbone networks, unfortunately, do not distinguish them from the best-effort services. Thus estimating the effective capacity meaning the traffic volume that the backbone networks maximally accommodate with keeping QoS guarantees for the services is very important for Internet Service Providers. This paper proposes a test-bed based on ns-2 to evaluate the effective capacity of backbone networks and then estimates the effective capacity of an experimental backbone network using the test-bed in terms of the service standard of the VoIP service. The result showed that the effective capacity of the network is estimated as between 12% and 55% of its physical capacity, which is depending on the maximum delay guarantee probability, and strongly affected by not only the type of offered workload but also the quality standard. Especially, it demonstrated that in order to improve the effective capacity the maximum end-to-end delay requirement of the VoIP service needs to be loosened in terms of the probability to guarantee.

• Journal of IKEEE
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• v.5 no.1 s.8
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• pp.8-15
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• 2001

This paper remarks IETF based requirements for IP/UDP/RTP header compression issued in 3GPP2 All IP Ad Hoc Meeting and protocol stacks of the next generation mobile station. All IP Network, for real time application such as Voice over IP (VoIP) multimedia services based on 3GPP2 3G cdma2000. Frames for various protocols expected in the All IP network Mobile Station (MS) are explained with several figures including the bit-for-bit notation of header format based on IETF draft of Robust Header Compression Working Group (ROHC). Especially, this paper includes problems of IS-707 Radio Link Protocol (RLP) for header compression which will be expected to modify in All IP network MS's medium access layer to accommodate real time packet data service[1]. And also, since PPP has also many problems in header compression and mobility aspects in MS protocol stacks for 3G cdma2000 packet data network based on Mobile IP (PN-4286)[2], we introduce the problem of solution for header compression of PPP. Finally. we suggest the guidelines for All IP network MS header compression about expected protocol stacks, radio resource efficiency and performance.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.176-179
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• 2003

The use of Multiparty Conference service based on SIP for VoIP provides is gradually magnified, and the work for continuous development and standardization on SIP is in the process of advancing. But, currently it is impossible for SIP to support identity discovery and distribution of each participant for multiparty conference. In this paper, we propose a SIP extension for guaranteeing security on the multiparty conference using SIP by adding new method and reconstructing header informations. With this, it is also possible to identify discovery and to distribute each participant using SIP extension when a call is established for closed multiparty conference.

• ETRI Journal
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• v.32 no.6
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• pp.843-853
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• 2010

A successful deployment of multimedia applications over wireless environments entails improving the quality of service (QoS), not only from a technical point of view, but also considering the quality of experience (QoE) from the final user's perception. Although objective QoE measure models avoid the difficulties of subjective surveys, subjective QoE assessments are essential to understand the way users evaluate the QoS. In this work, we study the effect of a wide range of parameters on the QoE of VVoIP applications in a real wireless scenario. Through a complete statistical analysis of users' ratings, we identify the following facts. Although the use of VVoIP in wireless networks does not yet represent an advantage for users, there are great expectations for all applications under study, and with greater popularity comes higher expectations. It is easier for respondents to identify good behavior than poor behavior. Whereas the respondents' frequency of Internet use does not impact on the scores, respondents' gender does. Finally, the most determining parameters of quality from a user's perspective were instability, video quality, voice distortion, usefulness, and graphical interface.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.4
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• pp.491-498
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• 2014

As internet services are being provided through a wireless access, the importance of quality of experience (QoE) is stressed that is defined as the quality that indicates user's actual feeling when a service is provided. Unlike quality of service (QoS) that can be expressed as a numerical value, it is difficult to represent QoE in an objective way. If an internet service is serviced over a wireless channel, its QoE can be affected by a number of factors such as fading, mobility and so on. This paper, therefore, attempts to specify the relationship between QoE and QoS by conducting practical measurements for the voice service through 3G high speed packet access (HSPA) access network. Analysing the measured results, it has been shown that received signal strength indicator (RSSI) has a great influence on mean opinion score (MOS) through transmission delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7C
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• pp.691-699
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• 2006

Despite of the in-depth effort to cantrol the variability in IP networks, quality of service (QoS) is still not guaranteed in the IP networks. Thus, it is necessary to deal with the audible artifacts caused by packet lasses. To overcame the packet loss problem, most speech coding standard have their own embedded packet loss concealment (PLC) algorithms which adapt extrapolation methods utilizing the dependency on adjacent frames. Since many low bit rate CELP coders use predictive schemes for increasing coding efficiency, however, error propagation occurs even if single packet is lost. In this paper, we propose an efficient PLC algorithm with consideration about the speech characteristics of lost frames. To design an efficient PLC algorithm, we perform several experiments on investigating the error propagation effect of lost frames of a predictive coder. And then, we summarize the impact of packet loss to the speech characteristics and analyze the importance of the encoded parameters depending on each speech classes. From the result of the experiments, we propose a new PLC algorithm that mainly focuses on reducing the error propagation time. Experimental results show that the performance is much higher than conventional extrapolation methods over various frame erasure rate (FER) conditions. Especially the difference is remarkable in high FER condition.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.875-879
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• 2002

CPL is a technique that serves various additional service in Internet telephony such as call forwarding, call blocking etc. IETF IPTEL working group developed this CPL standard. Users could request various additional services such as call forwarding, call blocking etc. by registering XML scripts to location servers. This paper would describe the design and the implementation skill of SIP proxy server that support these improved functionalities in detail. SIP registrar and SIP proxy server are designed and implemented in Linux platform because this platform serves fast and low cost development environment.

• Journal of KIISE:Computing Practices and Letters
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• v.11 no.4
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• pp.312-333
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• 2005

SIP(Session Initiation Protocol) has been proposed for session control protocol of Internet multimedia communication system like VoIP(Voice over IP). SIP has complicated session control steps to support various kinds of audio and video formats and to assure service quality of real time data communication. Up until now, existing protocol analyzers can not provide such detailed information of SIP based communication system. In this paper, therefore, we propose a new protocol analyzer as a tool that can analyze and diagnose SIP based multimedia communication system throughout the session initiation, data exchange and session change steps. The propose traffic analyzer, which is called STAT(SIP based Traffic Analysis Tool), Is implemented on Winder's environment so that it is generally usable and extensible. Since STAT analyze low level packets captured via Ethernet broadcasting property, it is able to provide session status and real time traffic monitoring information without any affection to the communication system. The STAT which is implemented in this paper. therefore, is expected to be a useful tool for developing and managing of a SIP based multimedia communication system.

• The Transactions of the Korea Information Processing Society
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• v.7 no.8S
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• pp.2722-2728
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• 2000

현재의 인터넷 망을 이용하여 음성, 화상 정보를 실시간으로 이용하고자 하는 다양한 응용이 시도되고 있다. 차세대 통신으로 주목 받고 있는 IMT-2000에서도 기존의 회선 교환망 대신 인터넷 망을 이용함으로써 경제성, 관리의 편의성, 새로운 서비스의 창출이 가능한 등의 이점이 있다. 인터넷 망이 최선의 노력(best effort)만을 제공하기 때문에 발생되는 신뢰성과 지연의 문제는 이미 많은 연구가 있어왔고 현재 어느 정도의 서비스 품질을 획득하여 VoIP(Voice Over Internet Protocol)와 같은 서비스가 실제로 이용되고 있다. 그러나 무선 통신의 경우는 이에 더하여 무선 구간에서의 자원 할당의 문제가 남아 있다. 본 연구에서는 코어 망으로 인터넷 프로토콜을 사용하는 차세대 All-IP 망에서, 무선 이동단말 간의 멀티미디어 서비스가 가능하도록 효율적인 주파수 할당을 지원하는 QoS 관리자를 설계하였다. 제안한 QoS(Quality Of Service)관리자는 요구 대역폭이 다른 멀티미디어 호 요청에 대해 융통성 있는 주파수 할당이 가능하도록 대국의 QoS 관리자와의 협상을 통해 제한된 범위 내에서 서비스 품질을 조절하여 보다 많은 호 연결 요청이 성공할 수 있도록 한다.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.4
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• pp.185-194
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• 2010

In this paper, we propose an application layer-based vertical handover management protocol, called multihoming RTP (mhRTP), for real-time applications with seamless mobility across heterogeneous wireless access networks. The proposed multi-homing RTP provides a soft handover by utilizing multiple available wireless access network interfaces simultaneously. The newly available path is dynamically added to the ongoing session by the mhRTP session manager. Also the decision making of QoS-improving or QoS-guaranteed handover is possible based on the estimation of available bandwidth in each candidate network. The performances of the proposed mhRTP have been analyzed through a series of simulations on OPNET network simulator. From the performance analysis, we confirmed that the proposed mhRTP can provide QoS-guaranteed vertical handover with efficient session managements.