SIP协议在VoIP终端的设计和实现

SIP是下一代网络中的重要协议,而基于SIP协议的VoIP业务已经对传统话音业务形成了重要的威胁,并成为各大运营商竞争的重点业务之一.概述了VoIP业务的发展现状,介绍了SIP协议的实体、消息机制以及它所提供的业务,通过对SIP协议的原理及工作流程的分析,论证了其在实现一个VoIP系统中的优势,在此基础上设计并实现了一个基于SIP协议的VoIP终端.     

下一代VoIP协议探讨

我国VoIP已经到了需要有新的起色的关头.业务的个性化和多样性的特点是VoIP发展的考虑重点,选择合理的技术来支持是发展的关键,目前可用来实现VoIP的协议有多种,本文重点分析了各种VoIP协议的特点以及下一代VoIP需要重点考虑的问题。     

MCI选择爱立信推进融合网络战略

近日，爱立信与MCI公司双方就部署爱立信Engine下一代多业务网络解决方案达成了协议，该协议旨在把MCI位于美国的国际网关业务从传统的电路交换演进至电信级VoIP软交换。在美国推出爱立信的Engine下一代多业务网络解决方案是MCI融合网络战略的关键举措。这种电信级、标准化的VoIP软交换解决方案将使MCI能够通过其IP骨干网在全球传输语音流量，同时还可以扩展网络功能，并在全球范围内提供下一代多媒体业务。     

多媒体通信技术专题——下一代VoIP协议探讨

我国VoIP已经到了需要有新的起色的关头，业务的个性化和多样性的特点是VoIP发展的考虑重点，选择合理的技术来支持是发展的关键，目前可用来实现VoIP的协议有多种，本文重点分析了各种VoIP协议的特点以及下一代VoIP需要重点考虑的问题。     

基于并联监控的VoIP识别控制系统的设计与实现

VoIP技术的不断发展给传统电话业务带来了巨大冲击,电信运营商希望准确了解网络中存在哪些VoIP业务.文章针对VoIP业务的多协议以及串联控制模式的不足,研究VoIP业务的综合识别技术及并联控制技术,设计并实现了适用于多协议的VoIP并联控制系统,详细描述了系统功能、框架以及关健模块的设计与实现,最后对系统进行测试,验...     

VoIP体系协议的分析与研究

VoIP是一种在Internet网络上进行语音通信的新业务。H.323、MGCP、Skype、H.248、SIP是VoIP的重要协议。论文在分析这些协议结构的基础上,研究和对比了各种VoIP协议的使用特点,为架构不同的VoIP网络提出了协议选择建议。     

VoIP技术在城域网中的应用方案探讨

本文回顾了VoIP技术发展过程。并在对VoIP所使用协议进行分析的基础上。提出了在城域网中开拓VoIP业务较为典型的方案。     

在VoIP网关中实现高精度包同步

随着VoIP网关能够在POTS线或T1/E1中继器与IP网络之间进行业务的互操作,传真、MODEM和电话业务正越来越多地依靠VoIP技术,取代了传统的TDM设备.在部署VoIP网关时,发送和接收时序同步的机理经常被忽视.因而需要改进包同步以提高业务的可靠性和质量,这一点在传统的电路交换设备中是能够实现的.     

VoIP管制政策开放时机探析

VoIP业务在全球呈现快速发展的势头,美国、欧盟、日本等国家和地区都加快了发展步伐,著名的Sonus网络公司通过调查预言,在未来4～5年中面向商业用户和个人用户的VoIP承载的业务量将超过传统的电话服务,Analysys咨询公司则预测VoIP技术将在10年内取代PSTN网。在中国,IP电话通话时长保持高速稳定增长,在长途电话中的比重也同步提升,对传统电话的冲击越来越大。从技术发展趋势和消费者的呼声来看, VoIP业务影响深远,开放VoIP业务大势所趋。面对VoIP业务渐盛的态势,美国采取开放的政策,欧盟各国加以适度限制,日本则将其纳入传统电信业务管制范畴。尽管迄今为止,全球范围内尚没有国家或地区针对VoIP制订完善、全面的管制规则,但总的趋势是从管制层面上支持VoIP业务的发展。我国目前对VoIP采取严格管制态度,只有基础电信运营商有权经营IP电话业务。但是,开放或适度开放VoIP业务是必然的政策走向,我国选择何种时机调整VoIP业务管制政策,是值得深入探讨的问题。     

VoIP流量监测技术的研究与应用

目前用于VoIP的流量监测技术主要是基于SIP协议或者H.323协议,其对标准VoIP流量的监测率较高,但是对目前所存在的各种非法VoIP业务的监测能力欠佳。文章提出了一种基于协议分析、端口匹配、深度净荷匹配来监测VoIP流量的综合方法,并运用于实际的VoIP监控系统当中,通过在电信现场环境下的验证,最终表明该方法对VoIP流量的监测准确率较高。     

基于HFC网络的VoIP解决方案

简单介绍基于HFC网络的电路交换语音解决方案，着重论述基于HFC网络和DOCSIS系统的VoIP解决方案，根据SCN的VoIP试验情况，提出有线运营商部署VoIP业务平台时的难点和应该考虑的问题，最后指出有线运营商应该充分利用其网络资源，积极进行语音业务的拓展，推出有竞争力的语音产品，满足用户的需求。     

Power saving mechanism for VoIP services over WiMAX systems

The IEEE 802.16 system provides the power saving class (PSC) type II as a power saving algorithm for voice over internet protocol (VoIP) service, but it is not designed to consider silent periods of VoIP traffic. The main objective of this paper is to introduce a power conservation mechanism that combines power saving modes class I, class II and class III which is applicable to VoIP service with silence suppression which basically follows PSC II during talk-spurt periods but in silence periods it combine PSC I and PSC III. According to experimental results, more than 96 % power reduction can be achieved in mutual silence period by using the proposed VoIP power saving mechanism for VoIP services during silent periods with respect to the Quality of Services.     

Measurement Scheme for One‐Way Delay Variation with Detection and Removal of Clock Skew

One‐way delay variation (OWDV) has become increasingly of interest to researchers as a way to evaluate network state and service quality, especially for real‐time and streaming services such as voice‐over‐Internet‐protocol (VoIP) and video. Many schemes for OWDV measurement require clock synchronization through the global‐positioning system (GPS) or network time protocol. In clock‐synchronized approaches, the accuracy of OWDV measurement depends on the accuracy of the clock synchronization. GPS provides highly accurate clock synchronization. However, the deployment of GPS on legacy network equipment might be slow and costly. This paper proposes a method for measuring OWDV that dispenses with clock synchronization. The clock synchronization problem is mainly caused by clock skew. The proposed approach is based on the measurement of inter‐packet delay and accumulated OWDV. This paper shows the performance of the proposed scheme via simulations and through experiments in a VoIP network. The presented simulation and measurement results indicate that clock skew can be efficiently measured and removed and that OWDV can be measured without requiring clock synchronization.     

Voice over IP service and performance in satellite networks

Voice over IP services have emerged as a low-cost alternative to PSTN voice service, and an attractive solution for voice/data integration in public and private networks. Satellite systems, as an integral part of the global communications infrastructure, already have an increasing portion of their capacities used to carry data packets, and with their global coverage and reach to remote areas are well positioned to enable growth of VoIP services. For VoIP over satellite, several issues need to be addressed. These include transmission and quality of service issues, as well as service-related issues such as service creation and customization, support of IN and supplementary services, and seamless integration with the PSTN. While the service-related aspects of VoIP are common to both terrestrial and satellite networks, transport-related issues are different. This article reports performance results of laboratory experiments for evaluating VoIP over satellite under different link and traffic conditions     

VoIP业务QoS性能及其优化研究

简要介绍了VoIP传输的基本原理,对影响VoIP业务QoS性能的3个主要因素(时延、抖动和丢包)进行分析,提出了利用MPLSdiffserv awareTE(流量工程)集成模型进行端到端QoS性能优化的方法。MPLSdiffserv awareTE能够感知CoS(服务等级),并根据CoS细粒度来预留资源,在每个CoS级别提供MPLS容错机制,能够为VoIP业务提供低丢失、低延迟、低抖动以及确定的带宽服务,很好地满足服务质量要求。     

漫谈VoIP技术发展趋势——H.325:第三代多媒体系统

目前在Internet或IP网络上应用的VoIP技术主要是基于H．323或者SIP开发的。随着技术和需求的发展，VoIP要求能够同时提供话音、数据和视频等多种业务，向下一代网络NGN演进。为了更好地满足NGN的需求，弥补现有系统的不足，ITU提出了下一代多媒体系统H．325协议的概念，它的重点在于实现控制单元和服务单元的分离，更好地支持多种媒体编码协议的互通，提高系统的QoS以及安全性。H．325有望成为下一代VoIP技术的支撑协议。     

非法VoIP行为检测与管控方法的实现

由于IP网络的开放性和自由性,以及VoIP接入技术的简单易行,非法VoIP业务的管控工作具有重要意义。对VoIP业务监控主要可通过实时监测并分析网络协议,结合流量分析技术的方法,对采集来的流量进行VoIP业务的连接控制协议分析和数据流分析,检测出所有的VoIP业务流量,并判断是否合法。     

实施VoIP过程中的安全性考虑

随着 VoIP(Voice over IP)技术在全球电信市场的不断普及,VoIP的安全问题迫切需要得到解决,而这也是运营商开展业务的前提.VoIP安全可以分为业务的正常提供、业务内容的保密、呼叫者身份确认和系统安全等.文章分析了VoIP目前存在的安全威胁及其相应的防范措施.     

基于SIP协议的IP电话语音系统设计与实现

本文基于SIP协议的IP电话作为主要研究内容,探讨了IP电话的相关议、相关标准和关键技术,对SIP电话协议进行了研究分析,设计提出一套结构合理的VoIP系统,并对系统组成、系统流程等作了详细的规划,安装配置OpenSER服务器,并能够使用MySql数据库系统来存储用户信息,基于分层设计的思想设计了客户端软件,最后利用该软件进行了系统测试,结果表明系统性能优良。     

Evaluating the perceived voice quality on VoIP network using interpolated FIR filter algorithm

Voice over Internet Protocol (VoIP) is a popular communication service nowadays. VoIP reduces the cost of call transmission by passing voice and video packets through the available bandwidth for data packets through Internet protocol. The quality of the VoIP signal is degraded due to the various network impairments. The proposed scheme, interpolated finite impulse response, is implemented as post-processor after decoding the signal in VoIP system. The performance of the proposed scheme is evaluated for various network conditions. The results of the proposed scheme are measured with the objective measurement methods for signal quality evaluation. The performance of the proposed system is compared with the existing techniques for quality improvement in VoIP system. The results show much improvement in speech quality with the proposed scheme in comparison to other similar schemes.