• Title/Summary/Keyword: internet telephony

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A Study on the Performance Analysis and synthesis for a Differentiated Service Networks (차등 서비스 네트워크에 대한 성능 분석과 합성에 대한 연구)

  • Jeon, Yong-Hui;Park, Su-Yeong
    • The KIPS Transactions:PartC
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    • v.9C no.1
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    • pp.123-134
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    • 2002
  • The requirement for QoS (Quality of Service) has become an important Issue as real-time or high bandwidth services are increasing, such as Internet Telephony, Internet broadcasting, and multimedia service etc. In order to guarantee the QoS of Internet application services, several approaches are being sought including IntServ (Integrated Service) DiffServ(Differentiated Srvices), and MPLS(Multi-Protocol Label Switching). In this paper, we describe the performance analysis of QoS guarantee mechanism using the DiffServ. To analyze how the DiffServ performance was affected by diverse input traffic models and the weight value in WFQ(Weighted Fair Queueing), we simulated and performed performance evaluation under a random, bursty, and self-similar input traffic models and for diverse input parameters. leased on the results of performance analysis, it was confirmed that significant difference exist in packet delay and loss depending on the input traffic models used. However, it was revealed that QoS guarantee is possible to the EF (expedited Forwarding) class and the service separation between RF and BE (Best Effort) classes may also be achieved. Next, we discussed the performance synthesis problem. (i. e. derived the conservation laws for a DiffServ networks, and analysed the performance variation and dynamic behavior based on the resource allocation (i.e., weight value) in WFQ.

Design and Implementation of SIP Internet Call-setup System using Seven States (7가지 상태를 이용한 SIP 인터넷 전화연결 시스템 설계 및 구현)

  • Shin, Yong-Kyoung;Kim, Sang-Wook
    • Journal of KIISE:Computing Practices and Letters
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    • v.13 no.5
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    • pp.300-310
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    • 2007
  • The Session Initiation Protocol (SIP) is one of the major protocols used in call-setup over IP telephony. The SIP-signaled calls use many-sided states according to a request of user. In this paper, we suggest seven states and some events that help developers to design and implement new applications efficiently. And they enable an object-oriented design of the system. If you design the call-setup procedure only by the processing model suggested in RFC 3261 over commercial network, a fatal error may occur in the system because of heavy data traffic or unpredicted exception cases. However, according to the suggested seven states, if they are predefined events in the current system state, the standardized processing routine is executed. Otherwise, they can be processed by the exception routine in system. All event processing routines are designed and implemented using Finite State Machine (FSM).

Finding Smartphone's Factors which Affect Satisfaction or Dissatisfaction based on KANO Model (KANO 모델을 활용한 스마트폰의 만족 및 불만족 요인 분석)

  • Lee, Sang-Gun;Lee, Sin-Seok;Kang, Ju-Young
    • The Journal of Information Systems
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    • v.20 no.3
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    • pp.257-277
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    • 2011
  • The current study categorizes factors of smartphone into three, using KANO model: attractive factors which cause only product satisfaction, must-be factors for dissatisfaction, and one-dimensional factors for both. Based on it, it presents a new model for the effects that smartphone factors have on satisfaction or dissatisfaction. The purpose is to theoretically explain that smartphone factors on which companies and users place a high value can actually affect satisfaction or dissatisfaction. After choosing 15 factors out of 25 which had been selected through literature study, these were divided into attractive, must-be, and one-dimensional ones. 93 out of 109 questionnaires returned were used for analysis. After frequency analysis using SPSS were conducted on the surveys, the factors were grouped, based on KANO table. The grouping results are as follows. Attractive factors include 'expansion slots for external memory, battery desorption, brand awareness, mobile banking and internet telephony'. Must-be ones include 'multi-touch, information security, entertainment, information retrieval, location based service and SNS. Finally, 'screen visibility, size of internal memory, the amount of internal memory, battery life, and response to after-sales service' are classified as one-dimensional factors. A critical finding of this paper is that since the results are different depending on the operating system of smartphones, it must be taken into consideration in studies on smartphones. The wide and rapid spread of smartphones has changed people's lifestyle as well as business environment, which forces companies to compete with each other to adapt to the changed circumstances. In this competitive system, studies on smartphone factors of satisfaction and dissatisfaction are essential for firms to establish a new strategy. From this point of view, the present paper is expected to be a basic material for enterprises not only to develop goods and services that maximize customer satisfaction and minimize dissatisfaction, but also to establish the future business strategy.

Real-time Implementation of MPEG-4 HVXC Encoder and Decoder on Floating Point DSP (부동 소수점 DSP를 이용한 MPEG-4 HVXC 인코더 및 디코더의 실시간 구현)

  • Kang, Kyeong-ok;Na, Hoon;Hong, Jin-Woo;Jeong, Dae-Gwon
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.4
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    • pp.37-44
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    • 2000
  • In this paper, we described the real-time implementation effort of MPEG-4 audio HVXC (Harmonic Vector eXcitation Coding) algorithm for very low bitrates, which has target applications from mobile communications to Internet telephony, on current high performance floating point TMS320C6701 DSP. We adopted a hardware structure for real-time operation. In order for software optimization, we used C- and assembly-language level optimizations for time-critical functional codes. Utilizing the internal program memory of the DSP as the program cache, the internal data memory overlap technique and DMA functionality, we could get a goal of realtime operation of HVXC codec both at 2 kbit/s and at 4 kbit/s. For an encoder at 2 kbit/s, the optimization ratio to original code is about 96 %. Finally, we got the subjective quality of MOS 2.45 at 2 kbit/s from an informal quality test.

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A "GAP-Model" based Framework for Online VVoIP QoE Measurement

  • Calyam, Prasad;Ekici, Eylem;Lee, Chang-Gun;Haffner, Mark;Howes, Nathan
    • Journal of Communications and Networks
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    • v.9 no.4
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    • pp.446-456
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    • 2007
  • Increased access to broadband networks has led to a fast-growing demand for voice and video over IP(VVoIP) applications such as Internet telephony(VoIP), videoconferencing, and IP television(IPTV). For pro-active troubleshooting of VVoIP performance bottlenecks that manifest to end-users as performance impairments such as video frame freezing and voice dropouts, network operators cannot rely on actual end-users to report their subjective quality of experience(QoE). Hence, automated and objective techniques that provide real-time or online VVoIP QoE estimates are vital. Objective techniques developed to-date estimate VVoIP QoE by performing frame-to-frame peak-signal-to-noise ratio(PSNR) comparisons of the original video sequence and the reconstructed video sequence obtained from the sender-side and receiver-side, respectively. Since processing such video sequences is time consuming and computationally intensive, existing objective techniques cannot provide online VVoIP QoE. In this paper, we present a novel framework that can provide online estimates of VVoIP QoE on network paths without end-user involvement and without requiring any video sequences. The framework features the "GAP-model", which is an offline model of QoE expressed as a function of measurable network factors such as bandwidth, delay, jitter, and loss. Using the GAP-model, our online framework can produce VVoIP QoE estimates in terms of "Good", "Acceptable", or "Poor"(GAP) grades of perceptual quality solely from the online measured network conditions.

Design and Implementation of JAIN SIP-based Softphone Client (JAIN SIP 기반 소프트폰 클라이언트의 설계 및 구현)

  • Kim, Byung-Ho
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.12 no.12
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    • pp.2301-2306
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    • 2008
  • SIP(Session Initiation Protocol) has become an universal standard for multimedia communications for both wired and wireless networks since it has been adopted as a standard protocol for IMS platform in 3GPP standardization organization at November 2000. In this paper, we design and implement a SIP-based softphone client program which provides telephony service between internet users and a call center equipped with VoIP gateway. A softphone client based on PC-to-phone connection should guarantee to provide interoperability with various VoIP gateways and higher portability to be able to operate on different PC environments. The softphone client program in this paper has been developed with SIP 2.0 standard protocol to support interoperability and with JAIN SIP and JMF package to achieve higher portability.

A Flow Control Scheme for the QoS Improvement of Multi-Service using IPv6 Hop-by-Hop Option Header (IPv6 홉-바이-홉 옵션 헤더 이용으로 멀티서비스의 QoS 개선을 위한 플로우 제어 방안)

  • 이인화;김성조
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.2B
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    • pp.250-262
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    • 2004
  • In IPv6 environment, the Internet Telephony, VoD(Video on Demand) and high capacity file exchange service will be more increased than IPv4. Therefore, the strict guarantee of QoS based on End-to-End and differentiated quality control schemes are simultaneously required. This paper proposes the flow control schemes on IPv6 network that the traffic is identified by flow and the QoS of multi-service is improved by QoS information in IPv6 hop-by-hop option header. The object of flow control includes not only non-default QoS traffic, which uses the flow label, but also best-effort or encrypted traffic. Therefore, the guarantee of real-time service is strengthened and the flow, which abuses unnecessarily the network resources, is effectively controlled. Also, this paper proposes the mapping scheme between the flow and MPLS by reflecting the minimum change of the existed network resource and the status of backbone network of ISP(Internet Service Provider). In the simulation result, It is shown that the proposed scheme is effective in the side of QoS on real-time services and utilization of backbone resources.

Design and Implementation of Call Object Management mechanism for Customer Channel integration of Customer Relationship Management Environment (CRM 환경의 고객 채널 통합을 위한 콜 객체 관리 메저니즘 설계 및 구현)

  • Han, Yun-Ki;Koo, Yong-Wan
    • Journal of KIISE:Computing Practices and Letters
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    • v.13 no.7
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    • pp.520-533
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    • 2007
  • The CRM(Customer Relationship Management) is the business strategy model for higher profits and competitive power of the enterprise in a new business environment. The large-scale customer response service technique uses internet, e-mail, SMS (Short Message Service), Telephony service, DM(Direct Mail) by customer channel point. Recently, business model diversify for new contract and retaining existing customer to the effort for a profitable model of business. This paper is based on Avaya PDS(Predictive Dialing System) model for CRM bond center. If the number of "available" agents are less than the number of inbound channels, then there may be real-time response problems in PDS system implemented. The Organization cannot afford to have many agents in available mode because of the high cost of manpower. This paper provides two contributions to the study. First, we present Call Object Management Mechanism of Customer Channel integration for reduce outbound consulting and reduce CallBack data in the PDS. Second, we design and implement the proposed system. Our simulation results show analysis of old model and proposed model. The proposed model can be efficiently used in Large-scale CRM.

Image Contrast and Sunlight Readability Enhancement for Small-sized Mobile Display (소형 모바일 디스플레이의 영상 컨트라스트 및 야외시인성 개선 기법)

  • Chung, Jin-Young;Hossen, Monir;Choi, Woo-Young;Kim, Ki-Doo
    • Journal of IKEEE
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    • v.13 no.4
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    • pp.116-124
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    • 2009
  • Recently the CPU performance of modem chipsets or multimedia processors of mobile phone is as high as notebook PC. That is why mobile phone has been emerged as a leading ICON on the convergence of consumer electronics. The various applications of mobile phone such as DMB, digital camera, video telephony and internet full browsing are servicing to consumers. To meet all the demands the image quality has been increasingly important. Mobile phone is a portable device which is widely using in both the indoor and outside environments, so it is needed to be overcome to deteriorate image quality depending on environmental light source. Furthermore touch window is popular on the mobile display panel and it makes contrast loss because of low transmittance of ITO film. This paper presents the image enhancement algorithm to be embedded on image enhancement SoC. In contrast enhancement, we propose Clipped histogram stretching method to make it adaptive with the input images, while S-shape curve and gain/offset method for the static application And CIELCh color space is used to sunlight readability enhancement by controlling the lightness and chroma components which is depended on the sensing value of light sensor. Finally the performance of proposed algorithm is evaluated by using histogram, RGB pixel distribution, entropy and dynamic range of resultant images. We expect that the proposed algorithm is suitable for image enhancement of embedded SoC system which is applicable for the small-sized mobile display.

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