Supplementary services in the H.323 IP telephony network

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     

话音交换技术的发展趋势

电话交换已有100多年的历史,最早是人工交换,经历了步进制交换、纵横制交换发展到程控交换。随着互联网技术的发展,近年发展起来的数据网已经具备了提供话音业务的能力,现在正经历着传统电话网和数据网融合的变革。从IP电话发展起来的软交换技术提供了融合的方法,为电话网的又一次演进提供了技术基础。     

Technologies on the horizon-voice vs. data: an estimate of futurebroadband traffic

A number of data and other services are examined, and the total digital traffic generated, compared to voice traffic in digital form, is estimated. The total yearly traffic generated by telephone calls is estimated. The total traffic in digital form is calculated for a number of data services involving terminals and the transmission of text in digital form as alphanumeric characters. Included are results for video telephones, data file transfers, electronic newspapers, information-age labor force, motion picture distribution, network television distribution, data generated by or about people, newspaper and periodical composition and airline reservations. It is found that the total is less than one third of telephone traffic. It is pointed out that telephone traffic dominates because a single call generates a very large amount of data     

Voice QoS in third-generation mobile systems

We analyze voice quality and system performance in third-generation mobile communication systems. We argue that in these networks that are expected to integrate voice, data, and multimedia services, the transport network can no longer be considered as a lossless transparent traffic channel. Rather, proper dimensioning is needed even for plain voice services. Furthermore, the efficient utilization of the network resources shared by voice and data traffic requires sophisticated traffic management in the statistical multiplexing environment. The contribution of the paper is twofold: first, voice quality in the cellular transport network is analyzed and network dimensioning criteria are derived. Second, building on advances reported in the literature and taking into consideration current standardization activities, a joint performance model is established for the air interface (AI) and the transmission network (TN) allowing for the analysis of end-to-end service performance within a uniform framework. To demonstrate the applicability of this model, we point to some performance problems in the multiservice environment and suggest and evaluate traffic management actions to overcome these     

Speech and language processing for next-millennium communicationsservices

In the future, the world of telecommunications will be vastly different than it is today. The driving force will be the seamless integration of real time communications (e.g. voice, video, music, etc.) and data into a single network, with ubiquitous access to that network anywhere, anytime, and by a wide range of devices. The only currently available ubiquitous access device to the network is the telephone, and the only ubiquitous user access technology mode is spoken voice commands and natural language dialogues with machines. In the future, new access devices and modes will augment speech in this role, but are unlikely to supplant the telephone and access by speech anytime soon. Speech technologies have progressed to the point where they are now viable for a broad range of communications services, including: compression of speech for use over wired and wireless networks; speech synthesis, recognition, and understanding for dialogue access to information, people, and messaging; and speaker verification for secure access to information and services. The paper provides brief overviews of these technologies, discusses some of the unique properties of wireless, plain old telephone service, and Internet protocol networks that make voice communication and control problematic, and describes the types of voice services available in the past and today, and those that we foresee becoming available over the next several years     

论下一代网络

下一代网络的发展并不仅仅是带宽的增加，更重要的是数据流量在向媒体流倾斜。下一代网络的核心定位不是传递信息，而是娱乐、消费、享受的载体，是未来娱乐经济和体验经济的支柱。面对着消费大众翘首以待的大规模视频应用的巨大市场，传统的IT厂商束手无策，未来的视频媒体流是一个全新的市场，历史事实早巳反复证明：传送语音的电话网络并不依赖于过去的电报技术，传输计算机件的IP网络不依赖于过去的语音电话技术。同理，强大的视频媒体流网也并不依赖于过去的计算机信息处理和传输技术。     

Towards successful VoLTE and VoWiFi deployment: network function virtualization solutions’ benefits and challenges

The Recent decades have witnessed intensive efforts from operators to implement methods enabling better control over network utilization, service usage, and service monetization. Nevertheless, they report significant growth in Diameter signaling traffic, especially policy management signaling traffic. More specifically, operators offering long term evolution (LTE) data-only services and planning for a massive introduction of voice over LTE (VoLTE) and voice over WiFi (VoWiFi) services need to tackle the enormous growth in Diameter signaling traffic. The biggest challenge for those operators is to find an appropriate solution, scalable enough to handle the unpredictable growth of Diameter signaling traffic; as the VoLTE and VoWiFi services will reshape the landscape of LTE policies. Throughout this paper, we propose a network function virtualization (NFV) based model, mature enough to tackle the challenges of those operators planning to launch VoLTE and VoWiFi, without impacting existing services and without jeopardizing current revenues. In our approach we first used standard VoLTE and VoWiFi message flow and referenced users’ behavior; then we considered NFV architecture characteristics. We finally referred to the latest experiments and test results related to NFV maturity cycle.     

Low‐cost and high‐performance: VoIP monitoring and full‐data retention at multi‐Gb/s rates using commodity hardware

Voice over IP (VoIP) is increasingly replacing the old public switched telephone network (PSTN) technology. In this new scenario, there are several challenges for VoIP providers. First, VoIP requires a detailed monitoring of both users' quality of service (QoS) and experience (QoE) to a greater extent than in traditional PSTNs. Second, such a monitoring process must be able to track VoIP traffic in high‐speed networks, nowadays typically of multi‐Gb/s rates. Third, recent government directives require that providers retain information from their users' calls. Similarly, the convergence of data and voice services allows operators to provide new services such as full‐data retention, in which users' calls can be recorded for either quality assessment (call centers, QoE) or security purposes (lawful interception). This implies a significant investment in infrastructure, especially in large‐scale networks which require multiple points of measurement and redundancy. This paper proposes a novel methodology, architecture and system to fulfill such challenges, called VoIPCallMon, as well as the data structures and necessary hardware‐tuning knowledge for its development. As distinguishing features, VoIPCallMon provides very high performance, being able to process VoIP traffic on‐the‐fly at high bitrates, novel services and significant cost reduction by using commodity hardware with minimal interference with operational VoIP networks. The performance evaluation shows that the system copes with the VoIP load of real‐world operators. We further evaluated the system performance at a fully saturated 10 Gb/s link and no packet loss was reported, therefore demonstrating the potential of commodity hardware solutions. Copyright © 2014 John Wiley & Sons, Ltd.     

Network architecture and traffic transport for integrated wireless communications over enterprise networks

A novel network architecture based on the IEEE 802.6 metropolitan area networks (MAN) is proposed to integrate the wireless and wired segments of a regional enterprise network (REN) within a city. This architecture functions like a distributed switch for all types of services, reducing traffic congestion by sharing the high capacity link dynamically and facilitating signaling, mobility management, call processing and network management through its distributed functions, transport facilities and broadcasting capability. It also serves as a peripheral gathering network of REN traffic for transport over a wide area ATM/BISDN, enabling integration of an enterprise's regional networks into a global EN. Two major wireless applications, i.e., wireless PABX (WPABX) and wireless LAN (WLAN) are discussed to illustrate the advantages of this MAN‐based architecture. Although a REN is likely to support a wide range of different services, voice and data will continue to be the predominant traffic generated by WPABXs and WLANs, respectively, and are also representative of isochronous and asynchronous multimedia traffic carried by future wireless networks. We compare the traffic capacity of several voice transport alternatives under integrated (voice/data) network traffic with various data traffic loads, and study voice and data integration under three different integration schemes by simulations. Results indicate that the MAN‐based architecture is most effective employing queue arbitrated (QA) access for asynchronous traffic, pre‐arbitrated access for constant bit‐rate isochronous traffic, and the new reservation arbitrated (RA) access for variable bit‐rate isochronous traffic, under a scheme that permits full sharing between QA and RA traffic. This revised version was published online in July 2006 with corrections to the Cover Date.     

P2P流媒体业务技术分析与设计

基于对等网络(P2P)模式的业务和应用给传统互联网带来了巨大的影响,如何更有效地利用P2P技术来发展新的增值业务,是P2P业务开展各方迫切需要解决的问题。新兴P2P流媒体运营商、主流数字媒体运营商、电信运营商业务实现方案及部署思路各不相同。对于电信运营商,目前网络上P2P应用最成熟的模式为P2P直播,初期开展业务容易,用户接受程度较高。建议运营商结合自身的带宽资源对业务流量采用"疏"、"堵"、"用"相结合的手段,先期实现P2P直播业务,逐步尝试开展P2P下载和点播业务,同时加强P2P流媒体业务的安全保障系统方案及开展该业务可能的盈利模式的研究,使P2P流媒体业务成为的电信运营商转型和发展的重要支撑业务。     

Skype协议分析及流量识别

近几年来,P2P网络技术发展迅速,Skype是创建Kazaa的组织开发的一个基于P2P的VoIP客户端,用户可以用Skype通过互联网进行语音通话.本文通过抓取Skype的流量数据进行协议分析,主要关注PC2PC的登录/注销,文字通讯,语音通信,文件传输及PC2Phone等过程,进而总结协议特征,提出了一种基于协议分析的Skype流量识别方法,结果显示识别率达到95％以上.     

A Digital Telephone Set for ISDN

In addition to the present voice communications services, more diverse digital telephone services are in very strong demand. Among the critical system requirements for designing digital telephone sets are how to economically realize new telephone services and how to ensure the continuity of existing analog telephone services. This paper describes the system requirements from both the service and technical points of view in the design of telephone sets. Based on these requirements, it further describes a method of designing network interface and telephone function blocks. An outline of the configurations and characteristics of digital telephone sets which can be operated by remote power feeding is presented as a design example.     

城域数据网P2P流量优化方案

分析了P2P流量在城域数据网中的流向,并从运营商网络建设的角度,提出了P2P流量优化方案,包括BRAS节点的分布部署、IPv6地址的加速引入、分布式地址转换技术的应用和对P4P技术的关注,为运营商在智能管道时代在流量管理方面获得主动。     

带抢占功能的WiMAX QoS接纳控制技术研究

WiMAX技术作为一种实现宽带无线IP接入的典型方式,具有速率高、架构使用方便、支持移动性等优点。接纳控制是WIMAX系统的关键技术之一,对系统性能有着极其重要的影响。本文分析了WiMAX接纳控制的固有特性,针对现有的常规机制和借用机制的缺陷,提出了一种可以使高优先级的实时业务抢占正在进行的低优先级非实时业务的服务带宽的抢占机制,并进行了仿真实验。实验结果表明,运用抢占机制,高优先级的语音电话等实时业务能够优先被处理,从而避免了一些非实时的业务长期占用带宽,而实时的业务得不到服务的情况发生。     

Optical networks for local loop applications

Some of the options for optical technology within the local loop environment are examined. In particular, passive shared access networks have been considered in some detail. These networks show great promise for delivering existing telephone services to small to medium business customers (4-30 lines) economically by the early 1990s. Extending fiber to the home will also be possible by virtue of a similar passive network infrastructure for customers requiring new broadband services beyond the single telephone line. For one-line plain old telephone service (POTS) customers, an intermediate approach of terminating the fiber network at the final network distribution point, with copper retained for the final leg, may be used prior to the provision of broadband services. A key feature of the passive optical network architecture is the use of wavelength-division multiplexing (WDM) as an upgrade strategy, allowing graceful upgrading from telephone services to multichannel high-definition television (HDTV) on gigabit/second bearers and full two-way switched broadband services employing wavelength routing across the network     

QoS support for integrated services over CATV

Cable TV has emerged as a promising access network infrastructure for the delivery of voice, video, and high-speed data traffic. A central issue in the design of protocols for CATV networks is to support different levels of QoS for diverse user applications. While CATV service providers and equipment have standardized, in the so-called MCNS protocol, the basic network architecture and interfaces, issues in the MAC layer for QoS support are likely to be left for differentiation in vendor products. This article first presents an overview of the basic CATV network architectural assumptions and the set of QoS requirements for supporting integrated services over CATV. It then discusses a MAC layer scheduling protocol that can efficiently multiplex constant bit rate traffic, such as voice over IP with guaranteed delay bound, and best-effort traffic, such as data services with minimum bit rate guarantee, while achieving fairness on any excess available bandwidth. The performance of this algorithm is illustrated by simulation results using Opnet. We also discuss a dynamic polling mechanism that enhances the link utilization while preserving delay bounds for latency-critical traffic     

A spread‐slotted random‐access protocol with multi‐priority for personal and mobile communication networks carrying integrated traffic

This paper describes and analyzes a medium access protocol for wireless personal communication networks serving integrated traffic. A multipriority mechanism, which is implemented and controlled explicitly by the users and without any involvement from the network is incorporated into the protocol. We examine the proposed protocol under reservation and nonreservation policies and evaluate its performance when handling integrated voice and data traffic. The proposed protocol is suitable to be used in Personal Communication Systems aiming to provide integrated (voice/data) services.     

基于位置的流量经营实现方案分析

话音经营时代,运营商纷纷推出了分区话音计费业务.流量经营时代,基于位置的流量营销将具有更广阔的市场空间.本文基于运营商的智能网络架构,详细研究了基于位置的流量经营所需的网络侧关键技术:用户位置获取和基于位置的差异化计费,以期对运营商开展位置流量业务提供参考.     

电力程控交换中心软交换组网设计

近年来,我国的电力企业取得了卓有成效的发展成就,电力通信业务突破了传统程控语音通信等单一的业务模式,逐渐朝着网络融合方向发展.传统的程控交换技术仅能够提供语音通话服务,难以满足多样化综合业务需求,因此,必须探索新的能够实现对电话交换网管理的新型通信技术.此次研究将宽带IP技术作为基础,结合最新通信技术软交换系统,在信令控制及媒体处理技术支持下,构建了电力程控交换中心软交换组网系统平台,为企业生产及通信业务发展提供帮助.     

Network Assisted Power Control for Wireless Data

The cellular telephone success story prompts the wireless communications community to turn its attention to other information services, many of them in the category of wireless data communications. One lesson of cellular telephone network operation is that effective power control is essential to promote system quality and efficiency. In recent we have applied microeconomic theories to power control taking into account notions of utility and pricing. Our earlier work has shown that this new approach to power control for wireless data performs better than traditional techniques applied for voice signals. However, the operating points of such a strategy result in an unfair equilibrium in that users operate with unequal signal-to-interference ratios. Further, the power control algorithms required to achieve such operating points are more complex than the simple signal-to-interference ratio balancing algorithms for voice. In this paper, we introduce a new concept, Network Assisted Power Control (NAPC) that maximizes utilities for users while maintaining equal signal-to-interference ratios for all users. The power control algorithm is easily implemented via signal-to-interference ratio balancing with the assistance of the network that broadcasts the common signal-to-interference ratio target.