3G手机VoIP话音的QoS实现技术研究

在对3G手机VoIP话音QoS的主要实现技术进行分析的基础上,提出了3G手机VoIP话音QoS新的实现技术。文中通过对实时传输控制协议（RTCP协议）的详细研究,同时根据3G系统无线信道的具体特点,说明了实时传输控制协议运用于3G手机VoIP话音的QoS控制中的缺陷,并阐述了相应的控制解决方法。在基于Android的3G智能手机的VoIP客户端软件中,综合运用VoIP话音QoS的主要成熟实现技术,同时结合文中提出的VoIP话音QoS的解决思路,实现了对VoIP话音的QoS的控制。基于Android的3G智能手机的VolP客户端软件通过在不同的网络环境条件下的测试,VoIP话音质量良好,说明文中提出的3G手机VoIP话音QoS新的实现技术具有一定的实用价值。     

VoIP技术--语音和数据的集成

介绍了VoIP(Voice over IP)的基本组成构件即网关(Gateway)、网守(Gatekeeper)的概念和用途。讲述了语音在IP网上传输的基本原理，语音和数据、数据和IP包之间的转换和传送过程。在H．323协议栈的基础上详细讲述了H．248，H．225，H．245等通信协议和语音编码G．729，G．23l，G．7ll等协议。最后对VoIP的语音服务质量作了较为详细的阐述。     

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。     

A call admission control framework for voice over WLANs

In this article a call admission control framework is presented for voice over wireless local area networks (WLANs). The framework, called WLAN voice manager, manages admission control for voice over IP (VoIP) calls with WLANs as the access networks. WLAN voice manager interacts with WLAN medium access control (MAC) layer protocols, soft-switches (VoIP call agents), routers, and other network devices to perform end-to-end (ETE) quality of service (QoS) provisioning and control for VoIP calls originated from WLANs. By implementing the proposed WLAN voice manager in the WLAN access network, a two-level ETE VoIP QoS control mechanism can be achieved: level 1 QoS for voice traffic over WLAN medium access and level 2 QoS for ETE VoIP services in the networks with WLANs as the local access. The implementation challenges of this framework are discussed for both level 1 and level 2. Possible solutions to the implementation issues are proposed and other remaining open issues are also addressed.     

vGPRS: A Mechanism for Voice over GPRS

This paper proposes vGPRS, a voice over IP (VoIP) mechanism for general packet radio service (GPRS) network. In this approach, a new network element called VoIP mobile switching center (VMSC) is introduced to replace standard GSM MSC. Both standard GSM and GPRS mobile stations can be used to receive real-time VoIP service, which need not be equipped with the VoIP (i.e., H.323) terminal capabilities. The vGPRS approach is implemented using standard H.323, GPRS, and GSM protocols. Thus, existing GPRS and H.323 network elements are not modified. Furthermore, the message flows for vGPRS registration, call origination, call release and call termination procedures are described to show the feasibility of our vGPRS system.     

基于SIP协议的VoIP网关的设计和实现

在传统方式下,话音业务和数据业务是由两个互相独立的网络———公共电话交换网(PSTN)和数据通信网分别提供的。公共电话交换网基于电路交换的结构,这种技术保证语音有优良的品质,但同时也给PSTN带来线路资源分配方案的效率低和通话传真成本高的局限性。数据通信网则是基于包交换,由于采用了简单健壮的IP协议,使得其在这几年发展迅速。分析了目前在IP通信领域的两大主流协议H.323和SIP的特点和差异,给出了对支持SIP协议的VOIP网关的软硬件设计方案。     

基于H.323建议的VoIP安全机制分析

由ITU-T提出的多媒体通信协议H．323,目前广泛应用在电信、行业和企业VOIP网络中。本文主要分析了基于H．323的VOIP的网络结构、通信过程以及安全解决方案。     

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。     

VoIP over cdma2000 1xEV-DO revision A

In this article we analyze performance of VoIP services over 1xEVDO-Revision A (DO-Rev A) networks and show that high-quality VoIP with unconstrained mobility and high capacity can be achieved. Together with quality of service (QoS) requirements, we emphasize practical issues such as mobility, degradation of feedback-channel quality, and packet overheads. Novel techniques are presented for voice processing such as smart blanking and adaptive dejitter playback buffer with time warping. These techniques help to meet QoS constraints to achieve a circuit-like voice quality while improving overall capacity. Detailed end-to-end simulations are presented and system capacity is analyzed under the QoS and system stability constraints. We claim that DO-Rev A can provide VoIP capacity comparable to circuit-switched cellular CDMA technologies (e.g., IS-2000) and simultaneously carry significant amount of other types of traffic such as non-delay sensitive applications and downlink multicast.     

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。     

Performance evaluation of packet aggregation scheme for VoIP service in wireless multi-hop network

In a wireless multi-hop network environment, energy consumption of mobile nodes is an important factor for the performance evaluation of network life-time. In Voice over IP (VoIP) service, the redundant data size of a VoIP packet such as TCP/IP headers is much larger than the voice data size of a VoIP packet. Such an inefficient structure of VoIP packet causes heavy energy waste in mobile nodes. In order to alleviate the effect of VoIP packet transmission on energy consumption, a packet aggregation algorithm that transmits one large VoIP packet by combining multiple small VoIP packets has been studied. However, when excessively many VoIP packets are combined, it may cause deterioration of the QoS of VoIP service, especially for end-to-end delay. In this paper, we analyze the effect of the packet aggregation algorithm on both VoIP service quality and the energy consumption of mobile nodes in a wireless multi-hop environment. We build the cost function that describes the degree of trade-off between the QoS of VoIP services and the energy consumption of a mobile node. By using this cost function, we get the optimum number of VoIP packets to be combined in the packet aggregation scheme under various wireless channel conditions. We expect this study to contribute to providing guidance on balancing the QoS of VoIP service and energy consumption of a mobile node when the packet aggregation algorithm is applied to VoIP service in a wireless multi-hop networks.     

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.     

H.323系统中呼叫转移的研究与实现

呼叫转移的主要功能是将发往目的地的呼叫,转移到其他通信网络或终端上.在基于H.323系列协议的VoIP系统中,呼叫转移由网守实现,该功能有助于提高呼叫有效接通率和开展增值业务.针对H.323协议中的网守功能进行了研究,在此基础上提出两种新的呼叫转移设计方案,并对其实际应用进行了分析.实践表明该方案具有良好的工程应用前景.     

Performance Analysis of the ATM Adaptation Layer 2 (AAL2)

In this paper, we analyze the performance of AAL2 multiplexer for a continuous time Markovian arrival process. AAL2 CPS (Common Part Sublayer) packets are multiplexed in the AAL2 multiplexing queue and transmitted in the transmission queue. This tandem structure suggests that the statistics of AAL2 CPS requires at least 2 dimensional state space. Furthermore, from a network-level point of view, cell multiplexing and de-multiplexing procedures are repeated at each AAL2 switching node. That requires simple analysis model. To solve this problem, we reduce the state space by showing that the output process of multiplexing queue can be modeled with the Coxian distribution. We propose a single dimension analysis model of the CPS transmission queue. When AAL2 convey both real and non real time short packets, QoS management is a problem. This is because the QoS of real time as well as non-real time packets is measured using different metrics – delay and cell loss ratio respectively. Most previous work is concentrated around delay performance due to the real time applications getting the primary attention. From the direct comparison of delay and CLR performance, we show that delay constraint is the dominant parameter in QoS of AAL2.     

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     

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     

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。     

一种支持VoIP语音高优先级传输的WLAN系统设计

针对无线局域网（WLAN）多址接入的特点以及VoIP业务自身的质量要求,提出了一种设计方法,将上下行数据发送分离,支持实时业务的上下行对等传输,同时通过一种改进的早期随机检测（RED）算法,实现基于语音抖动参数的流量控制,为语音业务提供QoS保证。     

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。     

基于CS6220的VoIP终端的研究

以往的IP电话设备主要采用H．323协议实现，但是H．323协议实现过程的复杂，给实际使用造成很多的问题。提出一种基于CS6220的VoIP终端的实现方法，VoIP协议采用实现简单的SIP协议，同时也能够确保电话业务的服务质量，电话业务接口的技术性能指标满足相关规范要求；重点阐述了基于CS6220的VoIP终端的软硬件设计方法；电话接口的性能测试结果表明，所提出的方案在因特网运行良好。