一种动态时分窄带多业务接入新方案

如何在窄带低比特率链路上进行高效的语音数据等多业务综合接入,并保证语音等实时业务的质量是目前多业务接入的一个重点问题.本文提出的动态时分多业务接入方案解决了Digital Data Network (DDN)专线上窄带压缩语音,ADPCM语音,传真,以及数据的同时接入问题,有效地解决了DDN专线上多业务接入的质量保证问题,提高了链路利用率.本文对于该方案的性能进行了分析,并与目前基于IP的多业务接入方案进行了比较.本方案能够提供目前的VoIP方案下所无法提供的语音,传真业务的质量保证,在多业务的支持上比VoIP更加简单,更有吸引力.     

SDH向MS-OTN网络迁移方案探讨

分析了SDH网络面临的挑战及新业务承载需求，以“建、融、改”的三步骤建设MS-OTN（分组OTN）网络的思路，采用SDH和MS-OTN网络融合组网方案，解决SDH核心层承载能力不足问题，又借助SDH接入层实现初期MS-OTN网络的深度覆盖；通过细分SDH网络改造场景以及单站、子网等改造方案，探讨SDH网络分批、分层的退网方案，实现专线业务整体迁移及网络平滑演进。     

集团客户专线业务解决方案

本文通过分析集团客户专线需求,结合移动公司网络现状,提出了基于SDH/MSTP传送网,基于IP网和PON接入,以及基于IP网和以太网接入的三种集团客户专线业务解决方案,经过不同角度的分析比较,给出了建议。     

宽带多业务平台组网方案比较

1多网重叠建网模式现有的各种组网技术SDH、WDM、ATM、IP,可以综合利用、混合组网。1.1SDH ATM IP PSTN传统的多网重叠组网方案采用SDH ATM IP PSTN方式提供服务。在物理层组建SDH光纤环,提供50ms环保护功能和高带宽。一般来讲,骨干层SDH环为2.5/10Gbit/s,汇聚层SDH环为2.5Gbit/s/622Mbti/s,在接入层提供E1、E3、STM-1等TDM接口。在传输网之上组建PSTN和ATM/FR/DDN或ATM/IP网,提供语音交换服务和ATM/FR、N×64kbit/s专线、宽带IP服务。这种组网方式的缺点在于:誗网络设备众多,需要多个网管系统,建设和运营…     

基于光交网的MSAP设备部署研究

MSAP多业务接入平台以SDH技术为基础,运营商主要用于集团专线业务的接入。本文结合当前PTN技术和光纤网络的发展,提出了一种基于光交网的MSAP设备部署方案,采用双网双核心设备,同时支持SDH与PTN技术,实现两张传送网在接入层的融合,利用粗波分和单纤双向技术,提高了光纤资源的利用率。     

运营商三网融合接入控制方式探讨

1前言三网融合是电信网、计算机网和有线电视网通过技术改造,能够提供包括语音、数据、图像等综合多媒体的通信业务。各大运营商目前已经能够提供VoIP语音和高速互联网数据接入,三网融合在接入控制层面主要是考虑IPTV的接入方式以及如何在多业务接入模式下实现综合运营并保障各类业务的服务质量。     

浅谈分组传送网技术及其在广电网络的应用

本文主要介绍了分组传送网,它能提供大颗粒专线业务接入的网络技术,能与现有SDH/MSTP、OTN网络无缝连接,有效保护原始投资,且平台部署灵活,相较于传统SDH/MSTP接入方式,更能节约项目成本,提升市场竞争力。     

基于ZL50120的GEPON网络终端设备的研制

简要介绍1GEPON的网络宽带接入方式,此方式是目前实现语音、数据与视频三网合一的理想方式,对于语音接入主要有VoIP和CESoP两种.提出一种基于ZL50120的ONU语音接入实现方法,采用CESoP方式,能够确保电话业务的服务质量,电话业务接口的技术性能指标满足相关规范要求.重点阐述了基于ZL50120的ONU的软硬件设计方法;电话接口的性能测试结果表明,所提出的方案在GEPON运行良好.     

烽火网络专线接入解决方案

一、专线客户对接入网络的要求主要有高可靠性:大部分专线客户希望采用帧中继、V.35和SDH专线,尤其是基于SDH的专线,是目前的潮流。高安全性:高安全性意味着用户隔离、数据的防窃听、防篡改、防病毒等。扩展性:大客户的业务需求、带宽需求、接口需求和网管需求在与时俱进,要求大客户网络有扩展性和灵活性。综合接入:有相当多的客户,有语音和数据业务的综合接入需求。全程网管:全程网管意味着节省运维投入,是运营商降低成本的重     

烽火通信PDH/SDH融合解决方案服务大客户接入

烽火通信结合传统PDH和SDH接入方式各自的优势,推出了PDH与SDH完美融合的大客户接入解决方案,在SDH设备上嵌入PDH光分支盘与远端PDH设备对接,有效解决了PDH系统的统一监控,减少了设备转接点,降低故障率,大幅提高了运行的可靠性。烽火通信SDH与PDH融合的大客户接入解决方案继承了PDH价格低廉、部署迅速等特色,同时又融合了SDH组网灵活、运营管理方便的特点,成为性价比最优的大客户接入解决方案。更为重要的是该方案将PDH设备真正纳入传输网网管平台,降低运营维护成本,增强网络运行的可靠性,消除管理盲区,通过有效的资源调配…     

VoWLAN通信平台的设计与实现

VoWLAN也叫VoWiFi或者WiFi VoIP。它是基于无线网络技术和VoIP网络,是两者的有机结合。即是通过WLAN提供VoIP业务,使得终端用户通过WLAN拨打IP电话成为现实。本文提出了在基于Linux操作系统的SIP应用服务器及VoIP网关中,如何通过SIP信令和传统的PSTN数据通信线路与无线网络无缝连接方案,从而实现IP网络与传统电话间的实时语音通信、电话会议、语音信箱、视频通信、短消息、数据传输等业务。本设计已成功应用于某企业的实时语音通信平台,获得良好的效果。     

基于VoIP的车内话音通信系统的设计

文章对VoIP技术进行了研究,分析了VoIP的技术原理及与电路交换相比具有的优势,比较了VoIP两种体制ITU-U的H．323和IETF的SIP的优劣。在此基础上根据车内通信系统发展的现状提出基于VoIP的设计方案。给出了系统体系架构,以及话音综合接入设备的参考设计,为未来多业务终端接人的车内话音通信系统的应用提供了新思路。     

一种新的VoIP组网解决方案

随着NGN语音网规模增大,设备管理、网络安全等问题成为网络运行维护的难点和重点.该文在分析当前NGN语音网建设中面临的诸多问题基础上,分析介绍一种新的VoIP组网解决方案以及相关协议,并进一步阐述边界网关和接入网关对这种方案所需做的功能特性支持以及基于其上的扩展网管特性.     

A call admission control framework for voice over WLANs

In this article a call admission control framework is presented for voice over wireless local area networks (WLANs). The framework, called WLAN voice manager, manages admission control for voice over IP (VoIP) calls with WLANs as the access networks. WLAN voice manager interacts with WLAN medium access control (MAC) layer protocols, soft-switches (VoIP call agents), routers, and other network devices to perform end-to-end (ETE) quality of service (QoS) provisioning and control for VoIP calls originated from WLANs. By implementing the proposed WLAN voice manager in the WLAN access network, a two-level ETE VoIP QoS control mechanism can be achieved: level 1 QoS for voice traffic over WLAN medium access and level 2 QoS for ETE VoIP services in the networks with WLANs as the local access. The implementation challenges of this framework are discussed for both level 1 and level 2. Possible solutions to the implementation issues are proposed and other remaining open issues are also addressed.     

Contention‐free approaches for WiFi MAC design for VoIP services: performance analysis and comparison

The exponential growth in the demand of voice over internet protocol (VoIP) services along with the increasing demand for mobility in VoIP services has attracted great research efforts towards provisioning of VoIP services in IEEE 802.11‐based Wireless LANs (WiFi networks). We address one of the important research problems, namely, the quality of service (QoS)‐aware efficient silence suppression in the bursty voice traffic, for provisioning VoIP services in WiFi networks. The research works in the recent literature on silence suppression in voice calls have been surveyed categorising them on how the activity arrival is notified to the access point (AP). In most of the recent schemes, notification of uplink activity arrival is done through contention based medium access mechanisms such as the distributed coordination function (DCF). Contention‐based medium access causes non‐deterministic delays, therefore such schemes are not suited to voice traffic which require strict delay bound guarantees. This paper focuses on the schemes which do not use contention based approaches for silence suppression in voice traffic. Analytical performance evaluation and comparison of such schemes is carried out. Two very important performance metrics are modelled mathematically. One is the expected polling overhead time that the schedulers in these schemes can save per voice call during one voice activity cycle as compared to that in the round‐robin polling scheduler. The other is the expected unnecessary wireless channel access delay that a typical first talk‐spurt frame experiences due to the specific design of each scheme. The numerical results of this evaluation lead us to the conclusion whether or not and to what extent each of these schemes is viable. Copyright © 2008 John Wiley & Sons, Ltd.     

100+ VoIP Calls on 802.11b: The Power of Combining Voice Frame Aggregation and Uplink-Downlink Bandwidth Control in Wireless LANs

The bandwidth efficiency of voice over IP (VoIP) traffic on the IEEE 802.11 WLAN is notoriously low. VoIP over 802.11 incurs high bandwidth cost for voice frame packetization and MAC/PHY framing, which is aggravated by channel access overhead. For instance, 10 calls with the G.729 codec can barely be supported on 802.11b with acceptable QoS - less than 2% efficiency. As WLANs and VoIP services become increasingly widespread, this inefficiency must be overcome. This paper proposes a solution that boosts the efficiency high enough to support a significantly larger number of calls than existing schemes, with fair call quality. The solution comes in two parts: adaptive frame aggregation and uplink/downlink bandwidth equalization. The former reduces the absolute number of MAC frames according to the link congestion level, and the latter balances the bandwidth usage between the access point (AP) and wireless stations. When used in combination, they yield superior performance, for instance, supporting more than 100 VoIP calls over an IEEE 802.11b link. The authors demonstrate the performance of the proposed approach through extensive simulation, and validate the simulation through analysis.     

Efficient silence suppression and call admission control through contention-free medium access for VoIP in WiFi networks

In view of the rapidly growing trend of migrating customers from traditional wired phones to mobile phones and then to VoIP services in the recent past, there is a tremendous demand for wireless technologies to support VoIP, specially on WiFi technologies which have already matured commercially. This has put forth great research challenges in the area of wireless VoIP. In this article we have addressed two core issues, efficient silence suppression and call admission control, in QoS provisioning for VoIP services in WiFi networks. In this connection we present a QoS-aware wireless MAC protocol called hybrid contention-free access (H-CFA) and a VoIP call admission control technique called the traffic stream admission control (TS-AC) algorithm. The H-CFA protocol is based on a novel idea that combines two contention-free wireless medium access approaches, round-robin polling and TDMA-like time slot assignment, and provides substantial multiplexing capacity gain through silence suppression of voice calls. The TS-AC algorithm ensures efficient admission control for consistent delay bound guarantees and further maximizes the capacity through exploiting the voice characteristic so that it can tolerate some level of non-consecutive packet loss. We expose the benefits of our schemes through numerical results obtained from simulations.     

面向固定移动融合环境的P2PSIP系统设计

全业务运营是电信市场继语音和宽带接入服务之后的下一个增长点,而基于IP的融合有线网络和无线网络的语音服务则是全业务的重点之一。本文通过分析现有VoIP网络存在的问题以及固定移动融合网络环境下VoIP的特点,提出一种新型双层重叠网架构的P2PSIP架构,并阐述了新型架构的优点及双层重叠网之间的通信机制。新型架构能有效提高系统的安全性、健壮性和用户节点资源利用效率,更好的满足固定移动融合网络环境下VoIP对带宽、网络质量和安全性的要求。     

VoIP — The Access Dimension

For a given voice quality, VoIP does not offer a particularly efficient transport solution. This becomes a concern in any bandwidth-constrained part of a network, in particular, a copper-based, access network. VoIP access systems need to be carefully designed if they are to offer an acceptable combination of quality and efficiency. This paper identifies and quantifies the various issues which need to be addressed in undertaking such a design, and suggests possible solutions. The challenges of distributing VoIP within the customer premises are also addressed, with specific and different solutions being proposed for the business and residential environments.