局域网IP电话系统

局域网ＩＰ电话系统是在基于以太网技术的局域网上传输ＩＰ话音的系统。多ＩＰ电话系统不仅能２话音,而且能提供多种智能服务,完全可以取代传统的专用小交换机（ＰＢＸ）系统。如果把ＩＰ电话在骨干网上的应用和ＩＰ电话在局域网上的应用结合起来,将会使电话的应用更为完善,并能在骨干网和局域网上同时体现ＩＰ电话的优点。介绍了局域网ＩＰ电话系统听网络、话机、关守、 ＰＳＴＮ接入设备等组件所使用的技术及其功能。     

中国电信建成二期IP电话骨干网

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     

北电网络与神州数码签约吉通ATM多业务骨干网建设

吉通网络通信股份有限公司与北电网络、神州数码公司于近日签署合同 ,吉通公司将选择北电网络为其建设全国范围内的异步传输模式 (ＡＴＭ)多业务骨干网 ,而神州数码将作为系统集成商参与骨干网建设并提供相关技术及服务。新建成的吉通ＡＴＭ骨干网络将创造一个高容量的多业务数据网络平台 ,与吉通宽带ＩＰ网络互相配合补充 ,向国内、国际用户和合作伙伴提供多种数据业务。同时 ,该网络还将作为吉通公司ＩＰ电话网的传输承载网络 ,从而加强对ＩＰ电话的质量保证。新建成的吉通多业务骨干网由汇接层和区域层组成 ,在网络层次上各节点分为核心…     

局域网电话介绍

什么是局域网电话 局域网电话使用统一的网络通信设备和布线来传输话音和数据。在传统的ＰＢＸ（专用小交换机）系统中,话音呼叫通过与办公ＰＢＸ连接的一系列标准话音线路进入办公室,即通过一种专用设备在标准的电话配线上接收和疏导话音业务量。然而在局域网电话话音网络中,话音呼叫通过基于 ＩＰ的 ＰＢＸ进行接收和疏导,而这种 ＩＰ ＰＢＸ是与现有的数据网络连接的。ＩＰ－ＰＢＸ可以是一种独立的解决方案,也可以分解为几种分散的装置。 网关：网关是完成电路交换呼叫至分组话音转换重要任务以及其逆向过程的部件,网关有时与路由…     

《IP电话/传真业务总体技术要求》内容简介(上)

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...     

电信 联通 吉通 网通 移动——IP电话你选谁？

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…     

IP电话在HFC上的实现

符合ＭＧＣＰ体系的ＨＦＣ上的ＩＰ电话系统充分利用了ＣＡＴＶ网络资源，从根本上满足了大规模ＩＰ电话服务的需要，在对ＨＦＣ上的ＩＰ电话系统进行讨论的基础上，提出了能同时处理两路语音信号的基于Ｃａｂｌｅ Ｍｏｄｅｍ的ＩＰ电话适配卡设计方案。     

浅谈区域网电话

在传统的专用小交换机（ＰＢＸ）系统中,话音呼叫通过与专用小交换机连接的一系列标准话音线路进入室内,即话音信号通过一种专用设备在标准的电话配线上接收和发送。然而在局域网电话网络中,话音和数据使用统一的网络通信设备和布线进行传输,即话音呼叫通过基于 ＩＰ的专用小交换机（ＩＰ－ＰＢＸ）接收和交换,而这种ＩＰ－ＰＢＸ是与现有的数据网络连接的。局域网电话一般包括以下几个关键部分。 网关网关是完成电路交换呼叫与分组话音之间互相转换的重要部件,有时与路由器合并在一起。 路由器 路由器通常是现有数据网络的一部分,它…     

思科强调IP+Optical策略

“ＩＰ +Ｏｐｔｉｃａｌ既能为电信运营商实现智能网的ＩＰ服务 ,又能够突破带宽瓶颈。”思科系统公司亚太区光网络事业部总监ＩｄｒｉｓＴ．Ｖａｓｉ说 ,“目前服务供应商面临的首要挑战是 ,如何以新的方式提供盈利的服务。迅速发展的电子商务应用和内容服务所面临的一个主要障碍是带宽。除非城域网和骨干网上有足够的带宽 ,否则 ,服务提供商就不能在广域网上支持为局域网设计的应用。解决这一难题的首选方案是采用ＩＰ +Ｏｐｔｉｃａｌ策略。”思科公司凭借其在ＩＰ路由、多协议标签交换（ＭＰＬＳ）、光纤网络技术以及内容／服务智能化上…     

世界电信发展趋势

本文简要介绍了世界电信发展的特点，指出骨干网和接入网的发展趋势，说明三网融合势不可挡。随着因特网技术的发展，ＩＰ电话迅速崛起，向传统的电信经营者发起挑战。作者分析了ＩＰ电话火暴的原因、指出ＩＰ电话存在的问题。最后介绍了电子商务的发展概况、优点，阐述了实现电子需具备的环境。     

QoS enabled voice support in the next generation Internet: issues,existing approaches and challenges

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     

IP语音包的自适应编码和封装算法的研究

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。     

The IETF Internet telephony architecture and protocols

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     

一种局域网IP电话驱动卡的硬件设计

从IP电话的原理入手，介绍了一种局域网IP电话驱动卡的硬件设计和实现。该驱动卡基于ISA标准总线，作为普通电话机与电脑的接口卡，完成了利用普通电话机在局域网上进行通话的基本功能。     

局域网IP电话系统PSTN网关实现

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。     

Efficiency of packet voice with deterministic delay

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     

Unified Mobility Manager: enabling efficient SIP/UMTS mobile network control

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.     

Real-time voice over packet-switched networks

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     

A multiplexing scheme for H.323 voice-over-IP applications

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.     

Voice over Internet protocol and human-assisted e-commerce

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