Voice over IP signaling: H.323 and beyond

Signaling has been one of the key areas of Voice over IP (VoIP) technologies since inception. H.323 was the key protocol that allowed interoperability of VoIP products and moved the industry away from the initial proprietary solutions. Once the VoIP industry started maturing, some limitations of H.323 came to light. In this article we provide an overview of H.323, describe its capabilities, and discuss how its limitations are being addressed using the concept of gateway decomposition. We also discuss how H.323 can coexist with other protocols such as MGCP, H.248, and SIP which are attracting a lot of interest in the VoIP industry today     

VoIP网络中H.323与SIP的比较与互通研究

在VoIP网络中,H.323协议在SIP协议出现之前就已经得到了广泛使用,因此,要实现H.323协议和SIP协议的互通是当前需要解决的一个重要问题。通过简要介绍这两种协议的体系结构,进一步分析互通过程中需要处理的主要问题,提出了实现H.323与SIP互通的网络结构模型,同时对互通所必须的信令网关进行了初步研究,从而解决了两种协议之间的地址转换与映射、消息转换与映射、媒体能力协商等。经实践证明,该互通方案是可行的。     

VoIP信令协议SIP和H.323的比较

VoIP系统凭借其低廉的话费和较好的语音质量,已经成为重要的电信业务,并有取代传统长途业务的趋势.许多组织研究并制定了IP网络上呼叫的协议标准,但有两种IP电话信令和控制标准最具有影响力.一种是ITU推荐的H.323协议,另一种是IETF的SIP.这两种协议代表了解决同一问题的两种不同的方法:H.323是信令基于ISDN Q.931和早期推荐的H系列协议的传统的电路交换的方法,而SIP是一种支持基于HTTP的IP网络的超轻量协议标准.本文,我们主要针对SIP和H.323的体系结构,可靠性,复杂性,可扩展性,可伸缩性以及支持业务类型方面进行比较.     

SIP和H.323协议互通的研究

下一代网络是当前的研究热点,H.323和SIP是VoIP网络的两大主流技术,基于它们的网络间互联互通是一个亟需解决的问题。文章从协议角度提供一个从SIP网络到H.323网络的互通单元(IWF),重点介绍互通单元的结构和功能模型。通过对H.323与SIP协议的分析比较,指出互通过程中需要处理的主要问题,实现网络寻址、地址转换和消息映射的基本方法,最后给出实现互通最典型的通信流程,为实现基于软交换的下一代网络提供参考。     

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系统中NAT穿越技术的研究与实现

基于会话初始化协议（SIP）的VoIP系统在Internet上已经取得了广泛应用,但在目前的实际网络环境中,由于大量NAT设备的存在,使对等网络（P2P,Peer to Peer）之间的呼叫和数据通信难以实现。分析了四种NAT的类型特点,介绍了现有的NAT穿越方法,提出了一种基于STUN与TURN方式相结合的实现各种NAT穿越的VoIP系统设计方案。该方案对SIP信令采用可靠的TCP传输方式,对流媒体数据采取最大交付的UDP传输方式。经过校园网内部之间的网络环境测试,该方案达到了很好的接通率。     

SIP and H.323 — Interworking VoIP Networks

H.323 is currently the industry's predominant standard for offering multimedia services over IP networks, although the session initiation protocol (SIP) continues to generate significant interest in the market-place for rendezvous-based real-time applications. With its greater maturity, H.323-based equipment continues to be deployed in operational networks. However, with the advent of SIP-based third generation mobile networks and the continued development of voice over cable TV networks based upon SIP, significant deployments of SIP-based networks are on the horizon. Interworking between SIP and H.323-based equipment is of significant importance to BT and its Joint Venture partners. This paper highlights some of the issues surrounding protocol interworking between the H.323 and SIP protocol sets.It can be shown that, while possible, interworking inevitably results in sub-optimal operation and a loss of features. Thus, as interworking is inevitably going to be required for a long time to come, and the result is less than perfect, the standards bodies should be encouraged to develop service-enabling extensions to their protocols that are compatible between both protocol suites wherever possible.     

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。     

Performance study of link layer and MAC layer protocols to support TCP in 3G CDMA systems

Providing support for TCP with good quality link connection is a key issue for future wireless networks in which Internet access is going to be one of the most important data services. A number of schemes have been proposed in literature to improve the TCP performance over wireless links. In this paper, we study the performance of a particular combination of link layer protocol (e.g., radio link protocol or RLP) and MAC retransmissions to support the TCP connections over third generation (3G) wireless CDMA networks. We specifically investigate two metrics - the packet error rate and the delay provided by RLP and MAC retransmissions - both of which are important for TCP performance. For independent and identically distributed (i.i.d) error channels, we propose an analytical model for RLP performance with MAC retransmission. The segmentation of TCP/IP packets into smaller RLP frames, as well as the RLP buffering process, is modeled using a Markov chain. For correlated fading channels, we introduce an analytical metric called RLP retransmission efficiency. We show that: 1) the RLP frame size has significant impact on the overall 3G system performance, 2) MAC layer retransmissions significantly improve the TCP performance, and 3) the RLP retransmission scheme performs better in highly correlated channels, while other scheme performs better in low correlated channels. Simulation results also confirm these conclusions.     

Optimization of Mobile IPv6-Based Handovers to Support VoIP Services in Wireless Heterogeneous Networks

The support of voice over Internet Protocol (VoIP) services in next-generation wireless systems requires the coupling of mobility with quality of service. The mobile node can experience disruptions or even intermittent disconnections of an ongoing real-time session due to handovers. The duration of such interruptions is called disruption time or handover delay and can heavily affect user satisfaction. Therefore, this delay needs to be minimized to provide good-quality VoIP services. In this paper, the focus is on the network layer mobility, specifically on mobile Internet Protocols (MIPs), since they are natural candidates for providing mobility at layer 3. Using analytical models, the authors evaluate MIPv4, MIPv6, fast MIPv6 (FMIPv6), and hierarchical MIPv6 (HMIPv6) and compare their performances in terms of handover delay for VoIP services. To optimize the handover delay, the authors propose to use the adaptive retransmission timer described in this paper. The results obtained using the adaptive timer technique show that for a 3% frame error rate and a 128-kb/s channel, the handoff delay is about 0.075 s (predictive) and 0.051 s (reactive) for FMIPv6. It is around 0.047 s [intra-mobile anchor point (MAP)] and 1.47 s (inter-MAP) for HMIPv6, around 1 s for MIPv6, and 0.26 s for MIPv4     

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。     

Voice over Internet protocol (VoIP)

During the Internet stock bubble, articles in the trade press frequently said that, in the near future, telephone traffic would be just another application running over the Internet. Such statements gloss over many engineering details that preclude voice from being just another Internet application. This paper deals with the technical aspects of implementing voice over Internet protocol (VoIP), without speculating on the timetable for convergence. First, the paper discusses the factors involved in making a high-quality VoIP call and the engineering tradeoffs that must be made between delay and the efficient use of bandwidth. After a discussion of codec selection and the delay budget, there is a discussion of various techniques to achieve network quality of service. Since call setup is very important, the paper next gives an overview of several VoIP call signaling protocols, including H.323, SIP, MGCP, and Megaco/H.248. There is a section on telephony routing over IP (TRIP). Finally, the paper explains some VoIP issues with network address translation and firewalls     

H.323和SIP在IP多媒体网络中互通的实现

随着IP电话和视频通信的发展,H.323和SIP作为IP多媒体通信领域中被广泛采纳的两种信令控制协议,受到业界的普遍重视。如何有效地实现这两种协议之间的互通,成为近年来国内外研究的热点。本文在简要分析H.323和SIP互通要求的基础上,提出了两者互通的实现方案,并对互通需要解决的关键问题进行了讨论。     

VoIP transmission mechanism based on TCP

As the widespread employment of firewalls on the Internet, user datagram protocol (UDP) based voice over Internet protocol (VoIP) system will be unable to transmit voice data. This paper proposed a novel method to transmit voice data based on transmission control protocol (TCP). The method adopts a disorder TCP transmission strategy, which allows discontinuous data packets in TCP queues read by application layer directly without waiting for the retransmission of lost data packets. A byte stream data boundary identification algorithm based on consistent overhead byte stuffing algorithm is designed to efficiently identify complete voice data packets from disordered TCP packets arrived so as to transmit the data to the audio processing module timely. Then, by implementing the prototype system and testing, we verified that the proposed algorithm can solve the high time delay, jitter and discontinuity problems in standard TCP protocol when transmitting voice data packets, which caused by its error control and retransmission mechanism. We proved that the method proposed in this paper is effective and practical.     

VoIP走势与难点

VoIP在全球的迅速发展不可逆转,但由于存在难于互操作的多种制式,协议体制标准化是其在全球业务市场上面临的主要问题。目前的趋势是MGCP或SIP与H.323结合,而MeGaCo信令协议进一步改善了互操作的效率。在业务拓展方面,在统一传信普遍应用的同时又发展了实时媒体通信——瞬息传信（IM）;另外还有移动VoIP。     

SIP-based VoIP network and its interworking with the PSTN

The Session Initiation Protocol (SIP) is considered to be a future powerful alternative to the H.323 standard as the signalling system for the dominant Voice over IP (VoIP) communications. This paper provides an in-depth analysis of SIP by describing the SIP protocol stack, summarising the main features of the protocol, and illustrating its architecture, message and operation. The paper also explains the architecture and the two key aspects of signalling interworking when SIP is interconnected with the PSTN.     

基于Asterisk的SIP/H.323视频网关研究

会话起始协议(Session Initiation Protocol,SIP)和H.323这2种协议都曾是主流的IP语音协议(Voice overInternet Protocol,VoIP),并且各自拥有大量的用户群。目前面对这2种协议的互通网关大多是以完成音频互通为目的,针对这2种VoIP协议之间的视频互通需求,提出了使用星号(Asterisk)IP专用分支交换(IP Private Branch eXchange,IP-PBX)软件将使用这2种协议的各类设备进行以视频为目标的互联互通方法,并描述了该方法实现的具体实现需要修改的内容和采用该方法实现二者互通时的呼叫流程。     

Evaluation of transport protocols for the session initiation protocol

SCTP is a newly developed transport protocol tailored for signaling transport. Whereas in theory SCTP is supposed to achieve a much better performance than TCP and UDP, at present there are no experimental results showing SCTP's real benefits. This article analyzes SCTP's strengths and weaknesses and provides simulation results. We implemented SIP on top of UDP, TCP, and SCTP in the network simulator and compared the three transport protocols under different network conditions.     

基于IP的最新视频通信技术及其应用

主要论述了基于IP的最新视频传输技术的概念、基本原理；通过压缩编码技术和IP网络传输来实现视频通信的形式、IP视频电话端到端几种方式及VoIP(Video over IP）网络的基本构成；构成VoIP网络各部分的设备的主要作用。重点讨论了最新视频传输技术的协议规范H．323标准和SIP标准及其实现与网络相关技术，并讨论了IP视频通信网的关键设备－－VoIP网关和VoIP的人机接口界面。