G.723.1 Codec在OpenH323协议栈上的实现

OpenH323是一个开放源码的VoIP(Voice over IP)协议栈,支持H．323和SIP等多媒体通信协议,为多媒体应用提供了一个很好的开发平台。G．723．1是ITU-T建议在中低速率多媒体通信中使用的语音压缩算法,目前该算法已在IP电话系统中得到广泛应啊。基于OpenH323协议栈实现G．723．1Codec有着十分重要的应用价值。介绍在OpenH323的软件终端上实现G．723．1Codec的基本方法,并可推广到G．729等其它多种语音压缩算法。     

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。     

浅析VoIP的通信协议及应用

阐述了VoIP发展的历史和基本原理,并分别介绍了IP电话的不同实现形式。此后介绍了VoIP技术中使用的主要通信协议,包括H．323协议族、SIP和网关分解协议。对各协议的体系结构和主要用途作了比较详细的介绍,并对三种协议的特点进行了比较和分析。根据VoIP的技术特点,最后介绍了VoIP技术在目前国内电信市场中的应用情况,并对其未来的发展作了展望。     

基于H.323的VoIP通信系统的安全性的研究

基于IP网络的语音传输(VoIP)技术目前已经发展成为一种专门的通信技术,随着IP电话技术的成熟和应用的普及,IP电话安全成为越来越迫切需要解决的问题。本文简要介绍了基于H．323协议的VoIP的基本原理及体系结构,全面的分析了其在安全方面存在的漏洞与不足,并从IP网络和IP电话产品本身两个方面提出了相应的安全解决方案,可以明显的改善其安全性能。     

VoIP软交换互通软件系统

IP多媒体技术的突破口是语音通信,其标志性应用就是IP电话,或称为VoIP(VoiceoverIP)技术。目前VoIP以基于H.323协议的电信级IP电话为主,由于H.323协议复杂,并将控制和承载功能合为一体,难以保证系统的可靠性和灵活性,很大程度上制约了系统的进一步发展;而基于MGCP(媒体网关控制协议)的IP电话则弥补了H.323协议的不足,因而得以迅速发展,而二者的结合与互通就显得尤为重要。     

精品出版快讯

VoIP技术与应用 人民邮电出版社书号:10830 定价:38.00元 本书主要介绍了电信运营级 VoIP技术的解决方案。内容包括: 如何使用IP技术传输语音,H.323 协议,会话初始化协议(SIP), 媒体网关控制和软切换体系结构, IP语音和7号信令系统,服务质 量等。为计划开发VoIP解决方案     

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。     

基于SIP协议的智能ATA的设计

VoIP电话以其独特的技术优势越来越得到通信领域的关注,目前基于 VoIP的基本通信协议有H．323协议、SIP 协议和MGCP协议3种。基于纯IP的 SIP借鉴了HTTP和SMTP,是一个应用     

MGCP在VoIP中的应用及优势

VoIP是IP网络之上的语音传输技术,基于软交换的VoIP不仅提供语音业务,还通过各种协议和设备给用户提供综合的数据业务。本文简单介绍了VoIP的主要协议之一媒体网关控制协议MGCP、当前广泛采用H.323的VoIP体系结构,以及采用MGCP的VoIP体系结构,详细分析了采用MGCP的VoIP体系结构较之H.323的VoIP体系所具有的优势。     

VoIP技术

VoIP是Voice Over IP的缩写，VoIP就是在IP网上打电话。IP网泛指基于IP协议的分组交换网络，包括internet和Intrannet。VoIP是一种通过对语音信号进行编码压缩处理后，将压缩帧转换成IP数据包在IP网络上进行传输从而达到语音通信的技术手段。由于IP网是采用统计时分方式分配和使用网络资     

民航甚高频VoIP新技术的应用研究与展望

VoIP(Voice over Internet Protocol)即网络话音通信,其工作原理是将模拟的声音数字化,经过压缩与封包之后,以数据包形式在IP网络实时传输。VoIP也叫互联网语音通信或IP电话。甚高频VoIP语音通信使用IP技术,在IP网络上布置支持数字音频的甚高频电台和内话系统,传输话音,区别于基于PCM(Pulse-code modulation,即脉冲编码调制)技术的传统数字基带信号传输。VoIP兼具操作功能性及灵活性,这是基于TDM的传统系统所不具备的。     

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.     

Implementing VoIP: a voice transmission performance progress report

Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VoIP. This article summarizes much of what we have learned in this work, and we hope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VoIP network performance.     

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。     

Anti-packet-loss joint encoding for voice-over-IP steganography

VoIP (voice over IP) is a kind of voice communication technology based on UDP/IP protocols. Packet loss will inevitably happen when the channel environment deteriorates, which can pose challenges to the reliable transmission of VoIP steganography. A steganographic model based on joint encoding was proposed. In this model, packet erasure coding was introduced to preprocess the secret data. And the encoded data were embedded into voice packets with minimum dis-tortion using matrix embedding. Furthermore, the influences of key parameters on the performance of joint coding were studied. The selection algorithm for optimal parameters was also given. Experimental results show that the proposed joint coding can effectively improve steganographic resistance to packet loss, and decrease the number of modifications to the voice stream.     

VoIP Performance over HSPA with Different VoIP Clients

High Speed Packet Access (HSPA) Holma H, Toskala A (in HSDPA/HSUPA for UMTS, 2006) is expected to provide enough bandwidth for voice over IP (VoIP) service. In this article we assess the performance of VoIP over HSPA with different VoIP clients and voice codecs. The simulations results show that VoIP can have a good voice quality over HSPA if a proper VoIP client and codec is used. However it is possible that the delay can increase with early HSPA implementations (mobile, network).     

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     

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。     

3G系统语音保密通信技术研究

通过对3G系统的技术标准的研究和对3G系统提供的相关业务流程的分析,讨论了通过3G系统进行语音保密通信的三种模式：即TrFO模式（Transcoder Free Operation）、类语音模式和VOIP模式。文中阐述了每种模式下保密语音的传输过程,利用类语音模式和VOIP模式,开发出的TD-SCDMA保密手机成功地实现了保密话音在3G系统（TD-SCDMA）中的传送,保密话音音质良好,而TrFO模式的实现则还有待于TrFO功能的开通。