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《通信世界》2000,(15)
本刊讯中国电信二期IP电话骨干网近日建成,并顺利通过原网IP话务割接工程。此项工程是由美国冠远科技股份有限公司(Clarent)承担的。 新建成的二期IP电话骨干网将代替原中国电信IP 电话实验网,覆盖全国范围。此次合作,冠远公司将为中国电信数据通信局的国内IP电话呼叫业务增添两项新服务。一项是拨打17900 接入的预付费IP电话服务。另一项是一步式服务,即中国电信现有的1.5亿部国内固定电话都可拨打17900按入 IP电话,呼叫费用记入主叫方的日常电话帐单中。 据分绍,1999年10月,中国电信已经采用冠远产品开通了预付费“宜通卡”服务,这是中国电信的国际直接电话(IDD)业务部的国际 IP电话业务,此次国内IP电话骨干网的建设进一步扩大了双方的合作范围。中国电信建成二期IP电话骨干网 相似文献
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《通信世界》2000,(17)
近日,IP标准研究组圆满完成了《IP电话/传真业务总体技术要求》的制定工作,经信息产业部批准,现已正式颁布发行。《IP电话/传真业务总体技术要求》对IP电话业务给出了全面而详细的规定,使国内IP电话标准有了一个明晰的概念,故此,本刊特分上、下两期对《IP电话/传真业务总体技术要求》作一个概要的介绍,以飨读者。 《IP电话/传真业务总体技术要求》(以下简称《总体技术要求》)是基于H.323协议系列,由于H.323是IP网会议系统的协议,不是专门为IP电话设计,再加上它原本设计是在局域网上使用,没有过多考虑到电信级网的特点,因而国内的技术体制在H.232基础之上作了必要的补充和增强。 1.IP电话业务网的体系结构 IP电话一般有四种组合形态,这里主要讨论IP电话中的两种组态,即电话到电话和PC到电话。在电信级的IP电话网中,尤以电话到电话组态为主。 在IP电话网中,用户电话的话音信息是经过两侧网关和IP网来沟通的。IP电话管理层的作用是用户认证、计费、地址翻译和网络资源管理等功能。对IP电话网来说,信息传递是一个平面拓扑结构,即语音流是在网关与网关间直接传递。电信级IP电话网的管理层则是分层的,分层结构的管理... 相似文献
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由于IP电话是经过一个IP电话网络来传输语音数据包的,因此 这个IP电话网络的好坏直接影响着IP电话的通话质量和服务水平。 IP电话网络类似于公路网,网络堵塞情形如同公路塞车。如果公路 路面很宽,路面质量很好,车子很多也不容易塞车。同样,如果IP 电话网络带宽很宽、容量很大,就可以容纳大量用户同时打电话 而不会造成堵塞。 目前国内的IP电话网络一般为公用互联网,骨干网带宽通常 只有几百兆,网络容量偏小,所以网络容量问题就成为直接左右 IP电话服务质量的重要因素。中国网通在去年10月底刚建成开通的 我国第… 相似文献
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QoS enabled voice support in the next generation Internet: issues,existing approaches and challenges 总被引:1,自引:0,他引:1
The Internet is under rapid growth and continuous evolution in order to accommodate an increasingly large number of applications with diverse service requirements. In particular, Internet telephony, or voice over IP is one of the most promising services currently being deployed. Besides the potentially significant cost reduction, Internet telephony can offer many new features and easier integration with widely adopted Web-based services. Despite these advantages, there still exist a number of barriers to the widespread deployment of Internet telephony. The most prominent one, however, is how to ensure the QoS needed for voice conversation. The purpose of this article is to survey the state-of-the-art technologies in enabling the QoS support for voice communications in the next-generation Internet. In this article, we first review the existing technologies in supporting voice over IP networks, including the basic mechanisms in the IETF Internet telephony architecture and ITU-T H.323-related Recommendations. We then discuss the IETF QoS framework, specifically the Intserv and Diffserv framework. Finally, we present two leading companies' (Cisco and Lucent) solutions to offering IP telephony services as examples to illustrate how real systems are implemented 相似文献
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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
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
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Internet telephony was first used as a simple way to provide point-to-point voice transport between two IP hosts. However, the growing interest in providing integrated voice, data, and video services has caused its scope to be extended. Internet telephony now encompasses a range of services, including not only traditional conferencing, call control, multimedia, and mobility services, but also new ones that integrate Web, e-mail, presence, and instant messaging applications with telephony. Internet telephony and traditional circuit-switched telephony will coexist for quite some time, requiring interworking between the two. In this article we present a suite of protocols, developed in the IETF, which provide a partial solution to this complex problem 相似文献
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郭会军 《电信工程技术与标准化》2003,(4):84-88
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
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Packet switching is appealing for carrying real-time traffic because it can benefit from (possibly variable bit rate) compression schemes and statistical multiplexing to more efficiently exploit network resources. This work explores the efficiency of IP telephony in terms of the volume of voice traffic carried with deterministically guaranteed quality related to the amount of network resources used. An IP network carrying compressed voice is compared to circuit switching carrying PCM (64 kb/s) encoded voice, and some design choices affecting IP telephony efficiency are discussed 相似文献
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Internet telephony is viewed as an emerging technology not only for wireline networks, but also for third-generation wireless networks. Although IP end to end is considered the ultimate approach to future wireless voice services, there is still a long way to go before IP voice packets can be effectively transported over the air. Therefore, Internet telephony and today's circuit-switched wireless network will coexist for years to come, and it is essential to effectively perform interworking between these networks. This article proposes the Unified Mobility Manager (UMM) that achieves efficient interworking between traditional wireless networks and Internet telephony networks. The main characteristic of the UMM is that it combines UMTS HLR and SIP proxy functionality in one logical entity, which helps eliminate the performance degradation due to interworking between SIP and UMTS. This article identifies seven potential network architectures with and without the UMM and with varying degrees of IP penetration in the wireless core networks, and performs comparative analysis in terms of their call setup signaling latency. Our performance results show that for SIP originated calls, the architecture with the UMM can achieve better performance than existing UMTS networks without the UMM. Our results further show that when the backbone network is fully IP-enabled, dramatic performance gains can be accomplished with the UMM for PSTN originated calls as well as for SIP originated calls. The article also demonstrates that the UMM allows graceful migration from today's circuit-switched wireless networks to hybrid SIP/circuit-switched wireless networks, and toward the IMS architecture for all-IP UMTS networks in the future. 相似文献
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We discuss the architecture and technical viability of transporting real-time voice over packet-switched networks such as the Internet. The value of integrating voice and data networks onto a common platform is well known. The telephony industry has proposed the ATM standard as a means of upgrading the Internet to provide both real-time and data services. In contrast, voice services may be added to traditional IP networks that were originally designed for data transmission alone. We consider the feasibility and expected quality of service of audio applications over IP networks such as the Internet. In particular, we examine possible architectures for voice over IP and discuss measured Internet delay and loss characteristics 相似文献
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A multiplexing scheme for H.323 voice-over-IP applications 总被引:1,自引:0,他引:1
Sze H.P. Liew S.C. Lee J.Y.B. Yip D.C.S. 《Selected Areas in Communications, IEEE Journal on》2002,20(7):1360-1368
Voice communications such as telephony are delay sensitive. Existing voice-over-IP (VoIP) applications transmit voice data in packets of very small size to minimize packetization delay, causing very inefficient use of network bandwidth. This paper proposes a multiplexing scheme for improving the bandwidth efficiency of existing VoIP applications. By installing a multiplexer in an H.323 proxy, voice packets from multiple sources are combined into one IP packet for transmission. A demultiplexer at the receiver-end proxy restores the original voice packets before delivering them to the end-user applications. Results show that the multiplexing scheme can increase bandwidth efficiency by as much as 300%. The multiplexing scheme is fully compatible with existing H.323-compliant VoIP applications and can be readily deployed. 相似文献
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Voice over Internet protocol and human-assisted e-commerce 总被引:1,自引:0,他引:1
By fostering the finalization of open standards and the convergence of voice, video, and data, the Internet protocol provides an ideal driver for the definition of the infrastructure for new multimedia and advanced communications applications. Voice over IP represents not only the chance to achieve cost-effective real-time voice communication over IP-based networks, but also the opportunity to build an integrated and open communications service delivery infrastructure. Developments of Web-based information systems and IP telephony in order to enable future e-commerce applications are summarized 相似文献