基于3GPP R7 HSPA的VoIP技术

基于HSPA的VoIP技术是目前3G研究一个热点问题,本文详细探讨了在3GP PR7 HSPA支持下的VoIP技术。研究结果表明．在同样端到端质量前提下．基于3GPP HSPA的VoIP频谱效率要高于基于电路交换的语音呼叫的频谱效率。这个较高的VoIP频谱效率主要是因为3GPP在R5／R6／R7规范中对高级移动接收机算法和VoIP优化无线网络算法进行了优化。研究还表明．即使HSDPA没有软切换．HSPA的移动性解决方案仍然可以满足VoIP的需求。     

VoIP over Wi-Fi Networks: Performance Analysis and Acceleration Algorithms

The expected VoIP call capacity in a one hop IEEE 802.11b network with G.711 voice codec is about 85 simultaneous calls, but the actual observed capacity is only 5 calls even at the highest data rate and under zero loss conditions. In this paper we analyze the reasons behind this inferior performance of VoIP traffic. We also present algorithms at the medium access control layer to improve the observed call capacity. Using ns-2 based simulations, we evaluate the algorithms and show that performance improvements of more than 300% can be achieved. Finally, using a testbed implementation of one of the proposed algorithms, we show its feasibilty in real world VoIP implementations. An earlier version of this paper was published in the IEEE Conference on Broadband Communications, Networks, and Systems (BroadNets) 2007 [1]. This work was supported in part by the National Science Foundation under grants CNS-0519733 and CNS-0721296, and the Georgia Tech Broadband Institute. Yeonsik Jeong was also supported under the Korea Research Foundation Grant KRF-2005-214-D00362 funded by the Korean Government (MOEHRD). This work was performed when Y. Jeong was a research scientist in the GNAN research group at Georgia Institute of Technology.     

3Sover HSPA技术研究

对CS over HSPA技术进行了详细的分析,阐述了该技术的基本原理、引入该技术后对3GPP原有标准的影响、对现有网络和终端设备的影响,并且重点对该技术的系统性能进行了分析     

HSPA＋无线网络性能与组网浅析

首先简要介绍了HSPA＋的高阶调制、MIMO、多载波等3种关键技术,然后从理论仿真的角度对64QAM、MIMO和DC3种技术的无线网络容量性能进行了分析,最后给出了HSPA＋网络推广的建议。     

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.     

无线VoIP与移动电话

由于当前的Wi-Fi技术多是基于LAN的,因此限制了Wi-Fi电话的应用。无线VoIP是VoIP与无线技术相结合的一种技术,具有价格便宜、应用范围广等优点,因此具有很大的发展潜力。通过对无线VoIP、普通VoIP和移动电话的比较分析,分别从他们的优缺点、安全性、呼叫质量、传输标准、用户群和价格等方面,探讨了目前人们比较关注的两种通信方式。     

VoIP Protection and Performance Improvement

A lot of ongoing work is geared towards improving the reliability, performance and QoS characteristics of service provider IP networks. In contrast, we propose novel enterprise-based techniques that exploit the fact that many enterprises are multi/dual-homed. In a form of service-aware routing, certain (e.g., VoIP) packets are duplicated (e.g., at one edge router) and sent over multiple service providers. After traversing the service provider networks, only the first-to-arrive packets are kept and the later-arriving copies are discarded. In so doing, the result is not only better protection against node and link failures, and packet losses and errors, but also better QoS performance under normal (fault-free) operation. The packet-duplication process can be policy-based and take into account costs, bandwidth, and priority issues, permitting the system to behave like a simple smart router that automatically and continually makes use of the best (lowest-delay) service provider. In this paper, we present the main ideas behind the proposal, along with some initial analytical and experimental/simulation results and insights from a Linux-based implementation.     

基于HSUPA的VoIP容量的研究

通过半解析方法预测和系统标准仿真两种方式,对基于高速上行分组接入(HSUPA)的VolP系统性能进行了研究.采用非调度算法、混合自动重传请求(HARQ)和软切换技术,降低了用户设备(UE)的功率,提高系统的容量.同时,鉴于VoIF业务对时间的敏感性,需在HARQ的最大重发次数和延迟预算之间取折中,以实现容量的最优化.结果显示,在VoIP中引入HSUPA后,容量比采用物理信道(DCH)的R99版本增加了12.2 kbit/s.     

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。     

使用跨Iur口策略提升HSPA数据性能

目前,高速上行行链路分组接入技术（HSPA）在数据速率方面的优势得到了用户的高度评价,但是在RNC边界上,数据速率的下降会严重影响用户感知。现对该现象进行了分析,提出了使用HSPA Over Iur策略的解决思路,并通过一个案例证明了该策略的可行性,该策略的使用不但可以解决边界数据速率的下降,而且减少了数据速率不稳定的问题,对提升3G数据性能提供了较好的参考。     

基于软交换技术的VoIP over HFC

本文对比了几种VoIP组网的标准,着重介绍了在有线电视HFC网上,基于PacketCable标准及软交换技术的VoIP解决方案,并对其中的关键技术进行了分析。     

HSPA+网络性能分析和部署探讨

主要从应用场景、基带资源分配、传输配置、载波策略等4个方面提出了HSPA+网络规划建议,并结合MIMO、DC-HSDPA技术提出了现网演进策略。     

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     

软交换技术在VoIP over HFC中的应用

介绍了VoIP电缆电话的发展现状,分析了各厂商软交换的试验和商业应用,结合长沙有线电视网络改造项目提出了实现VoIP over HFC的具体方案,最后就QoS和安全问题做了小结.     

Rice-Lognormal衰落信道中不同译码度量特性的比较

本文推导出了Ｒｉｃｅ－Ｌｏｇｎｏｒｍａｌ衰落信道中没有信道状态信息和具有信道状态信息两种情况下的最大似然译码度量,并同传统欧氏距离度量的错误概率进行了比较,数值计算结果表明：对于ＴＣ－ＭＰＳＫ方案,信道衰落特征（ｆＤＴ）对度量特性打电话几乎可以忽略。最后,讨论了Ｒｉｃｅ－Ｌｏｇｎｏｒｍａｌ衰落信道中的最小交织深度。     

HSPA＋关键技术及性能评估

就TD—SCDMA系统而言,HSPA技术有两种增强型方式：一种为多载波捆绑技术,另一种为HSPA＋。其中HSPA＋包括分组业务持续连接（CPC）、增强CELL—FACH、层2增强、下行64QAM和MIMO等5种子技术。本文将对HSPA＋涉及的各项关键技术进行详细分析,并对引入各项技术后带来的性能增益进行评估。     

Implementing a secure VoIP communication over SIP-based networks

Recent years the Session Initiation Protocol (SIP) is commonly used in establishing Voice over IP (VoIP) calls and has become the centerpiece for most VoIP architecture. As wireless and mobile all-IP networks become prosperous, free VoIP applications are utilized in all places. Consequently, the security VoIP is a crucial requirements for its adoption. Many authentication and key agreement schemes are proposed to protect the SIP messages, however, lacking concrete implementations. The performance of VoIP is critical for users’ impressions. In view of this, this paper studies the performance impact of using key agreements, elliptic curve Diffie–Hellman and elliptic curve Menezes–Qu–Vanstone, for making a SIP-based VoIP call. We evaluate the key agreement cost using spongycastle.jce.provider package in Java running on android-based mobile phones, the effect of using different elliptic curves and analyze the security of both key agreements. Furthermore, we design a practical and efficient authentication mechanism to deploy our VoIP architecture and show that a VoIP call can be established in an acceptable interval. As a result, this paper provides a concrete and feasible architecture to secure a VoIP call.     

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.     

P2P VoIP应用的性能评测

基于IP技术的语音分组传输(VoIP)电话目前被广泛使用,Skype与GTalk是VoIP应用的两个典型代表.在可控网络环境下,通过调整信道容量、时延、丢包、抖动等网络参数,利用PESQ MOS方法评测了Skype与GTalk的语音质量,并且讨论了在可变网络环境下的动态适应性策略.