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1.
Building blocks for IP telephony   总被引:3,自引:0,他引:3  
Convergence between the existing telephone networks and data transfer over the Internet not only demands that new software be written to handle telephony applications which span both networks, but also makes new and innovative applications possible. Rather than writing these applications from the ground up, it would be helpful to have a relatively high-level API on which to prototype new applications. In this article we describe a set of Java packages developed at Cornell University to accomplish this purpose. The software's name is ITX; it is available for download at no charge, and includes sample applications and documentation  相似文献   

2.
An analytical model for jitter in IP networks   总被引:1,自引:0,他引:1  
Traditionally, IP network planning and design is mostly based on the average delay or loss constraints which can often be easily calculated. Jitter, on the other hand, is much more difficult to evaluate, but it is particularly important to manage the QoS of real-time and interactive services such as VoIP and streaming video. In this paper, we present simple formulas for the jitter of Poisson traffic in a single queue that can be quickly calculated . It takes into account the packets delay correlation and also the correlation of tandem queues that have a significant impact on the end-to-end jitter. We then extend them to the end-to-end jitter of a tagged stream based on a tandem queueing network. The results given by the model are then compared with event-driven simulations. We find that they are very accurate for Poisson traffic over a wide range of traffic loads and more importantly that they yield conservative values for the jitter so that they can be used in network design procedures. We also find some very counter-intuitive results. We show that jitter actually decreases with increasing load and the total jitter on a path depends on the position of congested links on that path. We finally point out some consequences of these results for network design procedures.  相似文献   

3.
IP Telephony问题研究综述   总被引:3,自引:2,他引:3  
文中从对集成服务网络的主要应用之一分组网络语音传输应用的体系结构分析入手,从集成服务网络研究的观点出发,综述了以IP Telephony为代表的分组网络语音传输技术体系中上前尚有待研究解决的问题。通过文中的分析,我们讨论了电信网与因特网通过诸多技术方面的融合,走向合一网络的发展趋势。  相似文献   

4.
本提出在企业内部构建基于局域网的IP电话系统的迫切性和技术可行性,并结合目前最新的CTI技术,给出局域网IP电话系统重要组成模块——PSTN网关的一种具体实现方案。  相似文献   

5.
赵锋 《通讯世界》2003,9(5):48-52
目前,包括IP电话在内的动态应用还没有被企业广泛接受,其主要原因在于现有的网络基础设施,像防火墙和网络地址转换设备等静态网元不能有效地支持动态应用。安全IP电话的支撑部件(Secure Telephony EnabledMiddlebox,STEM)体系结构能够增强现有的网络设计并消除由于静态设备而带来的问题。目前,在数据网络上提供的业务已经从短的文本消息发展到了实时的语音和视频。这种转变要求在数据通路上采用更复杂和更智能化的网元。现有企业网通过大量的包括路由器、防火墙和网络地址转换设备(中间件)在内的静态设备连接在了一起。这些设备通过一…  相似文献   

6.
A new architecture that can be used for offering an Internet telephony service to residential customers is introduced. The architecture addresses scalability and availability requirements of mass-market deployment of carrier-grade services and supports interconnection with SS7 for Internet telephony calls to the public switched telephone network. The architecture is based on the concept of a gateway decomposition that separates the media transformation function of today's H.323 gateways from the gateway control function of the gateways and centralizes the intelligence in a call agent. The media gateway control protocol is introduced as the protocol between the call agent that assumes the gateway control function and the gateway that provides just the media transformation function. Interworking between the architecture and the public switched telephone network, the session initiation protocol, and H.323 are also discussed  相似文献   

7.
IP telephony has been rapidly introduced to replace the traditional circuit switched infrastructure for telephony services. This change has had an enormous impact on critical-infrastructure (CI) sectors, which are expected to become increasingly dependent on IP telephony services. Reliable and secure telephony service is a key concern confronting most organizations in the critical-infrastructure sector today. With the proliferation of voice over IP (VoIP) services in these organizations, it is important for them to understand the security vulnerabilities and come up with a set of best practices during the evolution of the IP telephony services. This article outlines the potential security issues faced by CI sectors as they transform their traditional phone systems into VoIP systems. Vulnerability analyses are conducted to understand the impact of VoIP security challenges in the new convergent network paradigm. The most common security measures are analyzed to identify their strengths and limitations in combating these new security challenges. A set of recommendations and best practices are offered to address the key issues of VoIP security as IP telephony is being introduced into critical infrastructure.  相似文献   

8.
Supplementary services in the H.323 IP telephony network   总被引:2,自引:0,他引:2  
Traditionally, different networks were developed to handle voice, data, and video. The circuit-switched telephone network carried voice and the packet network carried data. Due to different deployment of these networks, different services were developed, such as voice mail in the telephone network and electronic mail on the Internet. With the revolution of multimedia in the computer industry, voice, video, and data are now being carried on both networks. Supplementary services, such as transfer and forwarding (which were originally developed for private telephone networks and later migrated to public telephone networks) are now being developed for packet networks. The standards for packet networks are being defined in the H.323-based series of ITU-T recommendations. This article provides the H.323 architecture for supplementary services, the differences in deployment of these services between the circuit-switched and packet-switched networks, and interworking of these services across hybrid networks  相似文献   

9.
This paper describes a mixed-signal ASIC for dual-mode (analog/digital) cellular telephony applications. It consists of two transmit and two receive channels corresponding to the I and Q channels of a quadrature phase-shift keying (QPSK) modulation system. It also includes three 8 b DAC's for control purposes, as well as a bandgap voltage reference and bus interface circuitry. The chip is part of a four-chip implementation of an IS-54 dual mode telephone. The chip was implemented in a 0.8 μm n-well double-metal CMOS process and uses a 5 V power supply. The die area of the chip was 23 mm2 and the average power consumption was 125 mW  相似文献   

10.
针对近年来IP网络电话业务的大量涌现,简要介绍了IP网络电话合法侦听的概念,分析了与传统PSTN相比,对IP电话实施侦听的难点所在;接着设计给出了适用于H.323网络的、符合ETSI规范的合法侦听体系,并对比其他侦听体系分析了其优点所在。  相似文献   

11.
12.
13.
IP语音包的自适应编码和封装算法的研究   总被引:1,自引:0,他引:1  
黄永峰  李星 《电子与信息学报》2002,24(12):1829-1834
IP电话与传统电话相比语音质量较差,其中最主要的原因是因特网的带宽变化较大,导致丢包率较大。该文根据因特网带宽变化的特点提出了1种应用在IP电话网关中的语音自适应编码与封装策略,采用该策略的编码器能根据网络的带宽变化动态调节语音编码速率和语音包封装大小。据此,本文提出了4种算法:一种基于RTP协议语音包丢失率的计算算法、变速率编码算法,不同长度IP语音包的封装算法和根据丢包率来调整编码速率和封装的自适应算法。  相似文献   

14.
Seamless service delivery for mobile users complemented with Quality of Service provisioning for their real-time applications have a hot topic in the field of mobile communication in recent years. Seamless mobility goes hand in hand with Mobile IPv6 protocol. Since many different handover schemes trying to solve the Quality of Service issues have been developed a need for means for comparison has arisen. This paper presents an enhanced universal analytical method for comparison of handover schemes. The method focuses on two important aspects influencing the handover performance—binding update cost and packet delivery cost. The use of the proposed method is presented for comparison of four most common handover schemes—MIPv6, HMIPv6, FMIPv6 and F-HMIPv6.  相似文献   

15.
Discusses the integration of the telephone into the desktop work environment. This objective can be achieved in a number of different ways. The approach described uses the client-server model. Network telephony is a service that provides the capability to establish, answer, route, and terminate telephone calls under the control of applications on either desktop computers or servers resident in the network. It is accessed by applications via a standard programming interface (telephony services application programming interface, TSAPI) and utilizes a centralized server-based interface with the private branch exchange (PBX) to access the switching and telephone control services that the PBX provides. Creating a logical control link between the application software on the desktop computer and the telephone on the desk eliminates the need to physically connect the telephone to the desktop computer. The author distinguishes between the application programming interface and the network telephony service provider. The author assumes the services are provided by NTS R2.2 release and a PBX-based switching service. However, the author uses the term PBX to include switching services provided by key telephone or hybrid systems, PC-based telephony cards, or other appropriate technologies  相似文献   

16.
Many integrated circuits for large-scale application to telephone networks have been designed, including subscriber line interfaces, antialiasing filters, analog-to-digital-to-analog converters, and tone receivers. This paper summarizes in a unified fashion both interfacing and functional requirements for these devices, as well as the related circuit and technology approaches which have been utilized.  相似文献   

17.
A computer architecture for an accelerated, parallel, nondeterministic, discrete event simulator is described. The machine is evaluated for accelerating: road traffic simulation. The architecture employs reconfigurable logic, systolic arrays, and a reduction bus to perform microscopic discrete event simulation. The simulator, which achieves a speedup factor of at least 91 over its traffic software counterpart, is fast enough to be practical to municipal traffic management engineers handling road incidents in large metropolitan traffic networks  相似文献   

18.
Betts  J.A. Ghani  N. 《Electronics letters》1970,6(11):336-338
An adaptive version of the basic delta modulator employing full-width pulses and RC integration has been designed. A digital technique is used to sense the level of the input signal and to control the amplitude of the pulses supplied to the RC integrator in the feedback circuit. For an 800 Hz sinewave input a signal/quantisation-noise ratio of 32 dB has been obtained over a dynamic input range of 30 dB.  相似文献   

19.
The properties of noise fields in automobile interiors are discussed with a view toward speech enhancement for voice-activated mobile telephony. The limitations on performance of adaptive noise cancellation are explained in the context of the spatial correlation properties of the noise field. A simple delay-equalized near-field array of directional microphones is analyzed and found to be effective for increasing the signal-to-noise density ratio (SNR) and reducing the reverberant distortion of the speech, without introducing any further distortion. An array of N microphones, each with a delay and weight chosen according to its distance from the speech source, is a viable solution. Such an array gives gains on the order of N over the speech band, reduces reverberation, and does not introduce waveform distortion. Experimental results confirming the predicted performance are presented  相似文献   

20.
In this paper, we consider the evolution of telephone networks from time-division multiplexing circuit switching to packet switching and, in particular, to packet switching-based on Internet Protocol (IP-supported telephony). We analyze IP-supported telephony design solutions by proposing a layered reference model in which each layer is associated to a subset of the functions that support telephony. We use the reference model to establish a terminology and a framework for the comparison of the design solutions. We group the design solutions in scenarios and compare them in terms of the reference model proposed. We then focus on IP telephony, in which IP is used in telephone company networks, and on Internet telephony, in which the Internet is used to support telephony. We show that they both can be seen as implementations of the same architecture, which consists of a set of components, associated to functions, and of the interactions among these components. We then consider the issue of voice-data integration and analyze the variety of design solutions that can be adopted to integrate voice and data.  相似文献   

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