基于自适应变速率编码的VoIP网关的研究与应用

针对当前VoIP语音质量易受网络带宽影响的问题，提出了一种基于自适应变速率编码的VoIP网关。通过实时监测RTP语音分组的丢包率，并分析服务质量，自适应地选择最适合当前网络状况的编码速率，以提供一种语音质量和网络状况的最佳组合，从而降低语音分组的丢包率、有效地保障语音服务质量。结合开源项目Asterisk，实现了基于自适应变速率编码的VoIP网关。     

支持QoS的SIP代理服务器方案的设计与实现

从如何提高基于SIP协议的VoIP系统的语音质量出发，分析了该系统与MPLS技术结合处的难点，并提出了一种支持QoS的SIP代理服务器模型的解决方案。该方案的设计结构清晰、可扩展性强，解决了SIP代理服务器在IP网络中没有会话服务质量控制能力的问题，是一种好的融合VoIP及MPLS两大技术的应用模型。该方案目前已运用于基于嵌入式Linux平台的VoIP系统中，完全能够满足高质量语音的需求。     

大容量VoIP网关板设计及关键技术

随着IP语音业务的快速发展，研制高效、低成本且能节省IP地址资源的大容量VoIP网关成为新的课题，本文介绍了单板实现128通道的VoIP网关板的设计方案及关键技术，并对更大容量的VoIP网关设计提出了扩容方案。     

MIPS32 24KE内核设计应用于Ikanos公司的VDSL2住宅网关

MIPS科技宣布，Ikanos Cornmunieations，Inc．推出MIPS—Based Flusiv Vx180 VDSL2网关处理器。Ikanos的Vx180单芯片、多模VDSL2网关处理器旨在提供2．7GHz的处理能力、IP语音（VoIP）、多模DSL和安全性，同时支持一流的服务质量（QoS）和网速性能。目标应用包括用户终端设备（CPE）、家庭网络、SOHO及SME网络连接等新兴宽带领域中增长迅猛的领域。[第一段]     

提高VoIP语音通信质量方法的研究

探讨了VoIP的关键技术,分析了影响VoIP语音通信质量的因素,提出了提高VoIP语言通信质量的方法,即:优化网络环境、选择合适的编解码、服务质量保障(QoS)、使用颤音缓存。该方法对提高VoIP语音通信质量,推动VoIP业务的普及具有实际意义。     

VoIP网关中多链路SIP信令模块的设计与实现

文中在基于Linux操作系统的VoIP网关中，以单链路SIP信令实现为基础，提出了采用多进程架构的多链路SIP信令模块的设计实现方案。该设计继承了SIP协议固有的简单，可扩展性强等优点，而且多进程架构的设计结构清晰，能同时支持多路VoIP会话的建立与维护，解决了多用户的VoIP服务接入问题。该方案目前已用于基于Linux平台的VoIP网关的信令实现，完全满足VoIP网关的多链路处理与维护的需求。     

基于TCP友好控制机制的VoIP QoS研究

在基于UDP的VoIP应用中融入流量控制机制是确保其服务质量的重要方式。为了动态适应VoIP中的编码和分组，文中采用一种改进的TCP友好控制机制（TFRC）解决网络拥塞的问题，在端点对UDP语音流采用基于公式的发送速率调整，使之体现对TCP数据流的友好性。经过仿真和分析，该控制机制将公平地处理携带小分组的语音流和其他数据流，进一步保障了VoIP的服务质量（QoS）。     

嵌入式企业级VoIP网关的系统设计

通过Internet网络实现语音业务的传送能够有效地提高语音通信的经济性能，文章介绍了一种企业级VoIP网关的系统设计，以及如何用嵌入式的设计实现这样一个系统。描述了VoIP网关的硬件系统和软件系统的架构，讨论了它们的工作原理。     

基于NS-2的VoIP模拟与仿真

提出了一种基于自适应多速率算法的新VoIP系统方案,该方案可以根据网络信道质量的好坏来自适应地选择一种最佳的语音传输模式,从而在一定程度上保障了重构后的语音质量.在NS-2仿真平台下,通过与传统VoIP体系比较发现,所提出的新VoIP系统方案在语音QoS保障方面明要显优于传统VoIP系统.     

IEEE802.15.4无线接入VoIP网关的设计

提出了一种基于IEEE802.15.4无线个域网协议的无线接入VoIP网关的设计方案.该网关以FREESCALE MCF5234作为MCU控制芯片,使用MC13192作为无线收发器;整个软件体系通过模块化实现,结构清晰,易于扩展和升级,同时对无线网MAC层进行了优化设计,使无线收发稳定高效.该方案已用于基于eCos平台的VoIP网关的实现,完全满足两路语音的通话需求.     

Design and implementation of QoS-provisioning system for voice over IP

In this paper, we address issues in implementing voice over IP (VoIP) services in packet switching networks. VoIP has been identified as a critical real-time application in the network QoS research community and has been implemented in commercial products. To provide competent quality of service for VoIP systems comparable to traditional PSTN systems, a call admission control (CAC) mechanism has to be introduced to prevent packet loss and over-queuing. Several well-designed CAC mechanisms, such as the site-utilization-based CAC-and the link-utilization-based CAC mechanisms have been in place. However, the existing commercial VoIP systems have not been able to adequately apply and support these CAC mechanisms and, hence, have been unable to provide QoS guarantees to voice over IP networks. We have designed and implemented a QoS-provisioning system that can be seamlessly integrated with the existing VoIP systems to overcome their weakness in offering QoS guarantees. A practical implementation of our QoS-provisioning system has been realized.     

企业级VoIP媒体网关的设计与实现

企业级VoIP媒体网关是企业VoIP电话系统的一种解决方案,本文提出了一个企业级VoIP媒体网关的设计模型,重点描述了系统的功能其系统结构,并对系统各模块进行了详细的介绍,最后举例简要说明了该系统的呼叫流程。     

VoIP development in China

Several factors have spurred the explosive growth of VoIP phone use in China, including customer incentives such as improved voice quality and lower cost per call, and provider incentives such as higher profits and upgrade paths to next-generation technologies. The voice over Internet protocol, also called IP telephony, offers a new type of service that uses the Internet protocol, intranets, and extranets to deliver voice information. In contrast to traditional telephone services, which operate through a circuit-switched network, VoIP uses a packet-switched network. This distinction results in differences in implementation, quality of service (QoS), and operating costs. Since the service was introduced to the public in China in April 1999, VoIP toll telephone traffic has increased with astonishing speed. By the end of 2002, VoIP toll telephone traffic had surpassed traditional toll telephone traffic in China in both domestic longdistance and international call areas, including phone calls to and from Hong Kong, Macao, and Taiwan.     

基于H．323协议的企业级VoIP语音网关的设计和实现

讨论了基于H．323的企业级VoIP语音网关的设计原则，并根据该原则设计和实现了VoIP语音网关的基本模块。文章还简要介绍了H．323协议，同时对VoIP语音网关和VoIP技术作了小结。     

Blackfin平台上软件传真modem的设计与实现

在传统语音网关中集成传真功能是IP语音通信发展的必然趋势。分析了IP传真技术的研究和发展现状,在对传统语音网关的硬件平台不做任何改变的前提下,通过添加一个软件传真模块使其具备存储转发传真的功能,给出了该模块在Blackfin＋uClinux语音网关上的设计与实现。经过测试,该语音网关能够成功收发传真,并且性能良好。     

SIP的VoWLAN通信平台设计与实现

如何将会话初始化协议（SIP）与现有的通信网络有机结合,提供安全可靠的数据及通信服务已成为当今的热点问题。VoIP应用也受到业界的持续关注。安全问题一直都是企业实施VoIP的一个阻碍。提出了一个基于SIP的VoWLAN通信平台,将各种语音服务构建于无线局域网之上。利用虚拟专用网(VPN)、数据加密技术、VLAN和防火墙等必要安全技术和策略,应对在系统中的安全威胁,实现了通话质量可靠、安全性高的企业级VoIP无线网络架构。描述了该系统的设计和实现过程,讨论了其中的关键技术。     

Admission control by implicit signaling in support of voice over IP over ADSL

Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice over DSL. ATM is currently the preferred technology, since it offers the advantage of ATM’s built-in Quality of Service (QoS) mechanisms. IP QoS mechanisms have been maturing only in recent years. However, if VoIP can achieve comparable performance to that of VoATM in the access networks, it would facilitate end-to-end IP telephony and could result in major cost savings. In this paper, we propose a VoIP-based VoDSL architecture that provides QoS guarantees comparable to those offered by ATM in the DSL access network. Our QoS architecture supports Premium and Regular service categories for voice traffic and the Best-Effort service category for data traffic. The Weighted Fair Queuing algorithm is used to schedule voice and data packets for transmission over the bottleneck link. Fragmentation of large data packets reduces the waiting time for voice packets in the link. We also propose a new admission control mechanism called Admission Control by Implicit Signaling. This mechanism takes advantage of application layer signaling by mapping it to the IP header. We evaluate the performance of our QoS architecture by means of a simulation study. Our results show that our VoIP architecture can provide QoS comparable to that provided by the VoATM architecture.     

VoIP网关CID的设计实现

本文从如何实现VoIP网关中Caller ID同时在PSTN和Internet两个完全不同的网络传输的问题出发,介绍了PSTN中Caller ID的传输方式,分析比较了Caller ID在网络中传输的多种方案,并结合现有VoIP网关提出了一种基于SIP中SDP消息体的解决方案。该方案设计简单实用,目前已成功运用于基于嵌入式Linux平台的VoIP网关中。