RTP／RTGP和H．264在IPTV直播业务中的应用研究

本文旨在研究实时传输与控制协议（RTP／RTCP）在网络电视（IPTV）直播业务中的应用。首先分析了IPTV实时传输对通信网络的需求以及目前主要的通信网络对电视信息的实时传输的支持能力和存在的问题。然后着眼于IPTV网络直播业务,从网络协议出发,对适合音视频传输的网络协议TCP、UDP、RTP／RTCP进行了比较,分析了RTP／RTCP协议的优势以及不足之处,同时结合H．264视频编码标准探讨了QoS控制。最后展望了IPTV的发展前景。     

RTP／RTCP在IP-PBX系统中的应用

IP-PBX(IP-专用交换机)系统是近年来计算机网与传统电话系统技术融合领域的研究热点。而如何在IP-PBX系统中提供有QoS(服务质量)保证的实时通信能力成为一个关键技术。在介绍IP-PBX系统和RTP／RTCP(实时传输协议／实际传输控制协议)的基本原理和功能的基础上详细分析了利用RTP／RTCP为IP-PBX系统提供QoS保证的技术细节。     

基于RTP协议的网络流媒体组播传输技术

在简述RTP协议、组播技术及DirectShow技术的基础上,介绍了一个基于RTP协议的网络流媒体组播传输系统,重点对本系统的过滤器图的整体思路进行描述和演示.     

基于RTP协议的网络流媒体组播传输技术

在简述RTP协议、组播技术及DirectShow技术的基础上,介绍了一个基于RTP协议的网络流媒体组播传输系统,重点对本系统的过滤器图的整体思路进行描述和演示.     

RTP/RTCP和H.264在IPTV直播业务中的应用研究

本文旨在研究实时传输与控制协议(RTP/RTCP)在网络电视(IPTV)直播业务中的应用.首先分析了IPTV实时传输对通信网络的需求以及目前主要的通信网络对电视信息的实时传输的支持能力和存在的问题.然后着眼于IPTV网络直播业务,从网络协议出发,对适合音视频传输的网络协议TCP、UDP、RTP/RTCP进行了比较,分析了RTP/RTCP协议的优势以及不足之处,同时结合H.264视频编码标准探讨了QoS控制.最后展望了IPTV的发展前景.     

SRTP协议在MBMS中的应用

MBMS移动多媒体广播/组播业务,基于广播业务的一对多的特性,为了保证数据传输过程的保密性,流媒体传输过程的安全问题已受到普遍关注.一个具有安全保护功能的RTP协议即SRTP协议系统的实现不仅有利于对抗网络安全攻击,而且能够为用户提供隐私、加密和认证保护.SRTP协议通过对RTP数据分组中的有效载荷部分进行加密,从而实现RTP数据分组的机密保护.实验结果表明在MBMS中应用SRTP协议有效的保证了MBMS业务的机密性.     

RTP协议在嵌入式网络处理器中的设计与实现

分析了RTP/RTCP协议的特点与关键技术,设计了在嵌入式网络处理器中基于IP/UDP的RTP/RTCP协议的实现方法,并给出了在网络处理器IP2022中实现实时传输协议的主函数的流程图及局端与用户端的实现方法流程图.通过Commview软件的抓包实验验证了实现方法的正确性.     

IP视频会议系统中传输协议的应用浅析

主要介绍基于H．323／SIP的视频会议系统中,RTP／RTCP协议的概要和具体应用的几点分析。     

基于无线网络的实时视频传输系统的设计

通过对流媒体技术和网络协议的深入研究,设计和实现了一种基于RTP(实时传输协议)和FEC(前向纠错)的无线视频实时传输系统.该系统采用C/S(Client/Server)架构,服务器端基于RTP协议对数据包进行封装,并加入FEC纠错机制;客户端根据RTP数据包的头部信息,重建包序列,定位丢失的数据包,并利用FEC算法对丢失数据进行恢复.实际测试证明该系统能够在无线网络中有效地保证接收端数据的实时性和完整性.     

Linux下实时音频传输的实现

RTP是进行实时流媒体传输的标准协议和关键技术，文中介绍了RTP协议适合实时传输的特性及RTP数据包和RTCP控制包的格式，并简要介绍了Linux音频接口OSS的开发，最后给出了在Linux下利用RTP／RTCP进行实时语音传输的实现。     

VoIP电话的安全问题及防护措施

VoIP电话是综合了传统电信技术与计算机网络技术的一种新型应用。VoIP电话在传输时将信号压缩后封装成IP包，在IP网络上传输，这种传输方式存在着各种安全隐患。讨论了VoIP电话多种可能存在的安全问题，并就这些安全问题作了详细的分析，同时提出了防护措施，以最大限度地保障VoIP电话的安全。     

Vulnerability analysis and best practices for adopting IP telephony in critical infrastructure sectors

IP telephony has been rapidly introduced to replace the traditional circuit switched infrastructure for telephony services. This change has had an enormous impact on critical-infrastructure (CI) sectors, which are expected to become increasingly dependent on IP telephony services. Reliable and secure telephony service is a key concern confronting most organizations in the critical-infrastructure sector today. With the proliferation of voice over IP (VoIP) services in these organizations, it is important for them to understand the security vulnerabilities and come up with a set of best practices during the evolution of the IP telephony services. This article outlines the potential security issues faced by CI sectors as they transform their traditional phone systems into VoIP systems. Vulnerability analyses are conducted to understand the impact of VoIP security challenges in the new convergent network paradigm. The most common security measures are analyzed to identify their strengths and limitations in combating these new security challenges. A set of recommendations and best practices are offered to address the key issues of VoIP security as IP telephony is being introduced into critical infrastructure.     

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。     

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Trixbox的VoIP网络电话系统应用探讨

介绍了VoIP网络电话特点和应用现状,研究了Trixbox开源软件的结构、组成特点。设计了网络电话的系统结构,从服务器系统配置出发,重点讨论了VoIP网络电话系统图形化配置方案。针对IP网络电话安全问题提出了若干解决方案,对于搭建中小企业网络电话服务平台具有借鉴意义。     

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。     

Perceptual QoS assessment technologies for VoIP

Since quality is not generally guaranteed in an IP network, the proper design and management of networks and/or terminals for high-quality voice over IP services and maintenance of service levels is important. In terms of quality design and management, methodologies for appropriately and effectively evaluating the perceptual QoS of VoIP are indispensable. This article gives an overview of the state of the art of quality assessment technologies for VoIP, including recent work on improving their accuracy.     

VoIP over DVB-RCS with QoS and bandwidth on demand

Motivated by the need for compliance/interoperability above the satellite-specific layers, this article proposes a consolidated approach for voice over IP over satellite networks based on the ETSI DVB-RCS standard. Voice communication is a real-time service that needs priority over other services in IP environments with limited bandwidth, such as IP satellite networks. Bandwidth utilization in such networks needs to be optimized in order to reduce service costs, and this requires the use of dynamic bandwidth allocation schemes. This article therefore addresses the role of bandwidth on demand in the optimization of bandwidth allocation for VoIP and assesses the impact of BoD mechanisms on voice quality. The trade-off between voice quality and bandwidth efficiency is investigated under different DVB-RCS-specific capacity request/allocation strategies, and it is demonstrated that DVB-RCS provides an efficient platform for integrated support for a variety of VoIP applications over satellite. The main contribution of this article consists of the identification of the mechanisms capable of responding to the key challenges raised by the VoIP application in the satellite environment.     

IP电话及其发展方向

介绍了ＩＰ电话产生的技术背景和发展过程,描述了网关ＩＰ电话的实现方法,并对ＩＰ电话面临的问题、ＩＰ电话与传统电话的关系及发展趋势进行了分析和探讨。     

VoIP mobility in IP/cellular network internetworking

This article explores VoIP mobility in the context of IP and cellular networks interworking. ITU-T Rec. H.323 gateways provide the interconnection between IP networks and switched circuit networks. They allow a call originating from an SCN phone to be transmitted over an IP network to an H.323 terminal, or bridged to another SCN phone. While H.323 provides interoperability with other SCN terminals, the major efforts have been focused on IP/wired SCN (PSTN, ISDN, etc.) interworking. In this article we discuss the challenges associated with the interworking between IP networks and cellular networks through H.323 gateways, and propose an innovative approach using the existing call transfer supplementary service to provide VoIP mobility in the H.323 IP telephony networks. The proposed approach uses existing components in the H.323 standard, thereby allowing VoIP mobility service in hybrid IP/cellular networks to be a value-added feature in the existing H.323-compliant Internet telephony systems