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.     

网络流量识别方法研究

随着P2P和多媒体流量的发展,原有的流量识别方法越来越显现出其不足.为识别这些流量,需要识别效率更高的识别方法.文中提出了一种混合流量识别方法,此方法将特征识别和会话行为映射方法相结合,进行精确的流量识别.接着给出了这种识别过程的流程图,对识别过程进行了说明.对包的识别是基于优先级的特征识别,以提高识别效率.通过实验,选取四种应用Monkey3,eDonkey2000,MSN messenger,BitTorrent流量,将这些单个流量和混合流量的识别结果进行了比较.由实验结果可得出该方法对混合流量识别率比单个流量识别率高.     

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。     

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。     

Mobility management for VoIP service: Mobile IP vs. SIP

Wireless Internet access has gained significant attention as wireless/mobile communications and networking become widespread. The voice over IP service is likely to play a key role in the convergence of IP-based Internet and mobile cellular networks. We explore different mobility management schemes from the perspective of VoIP services, with a focus on Mobile IP and session initiation protocol. After illustrating the signaling message flows in these two protocols for diverse cases of mobility management, we propose a shadow registration concept to reduce the interdomain handoff (macro-mobility) delay in the VoIP service in mobile environments. We also analytically compute and compare the delay and disruption time for exchanging signaling messages associated with the Mobile IP and SIP-based solutions.     

The effect of packet dispersion on voice applications in IP networks

Delivery of real time streaming applications, such as voice and video over IP, in packet switched networks is based on dividing the stream into packets and shipping each of the packets on an individual basis to the destination through the network. The basic implicit assumption on these applications is that shipping all the packets of an application is done, most of the time,over a single path along the network. In this work, we present a model in which packets of a certain session are dispersed over multiple paths, in contrast to the traditional approach. The dispersion may be performed by network nodes for various reasons such as load-balancing, or implemented as a mechanism to improve quality, as will be presented in this work. To study the effect of packet dispersion on the quality of voice over IP (VoIP) applications,we focus on the effect of the network loss on the applications, where we propose to use the Noticeable Loss Rate (NLR) as a measure (negatively) correlated with the voice quality. We analyze the NLR for various packet dispersion strategies over paths experiencing memoryless (Bernoulli) or bursty (Gilbert model) losses,and compare them to each other. Our analysis reveals that in many situations the use of packet dispersion reduces the NLR and thus improves session quality. The results suggest that the use of packet dispersion can be quite beneficial for these applications.     

Mobility management for VoIP in 3G systems: evaluation of low-latency handoff schemes

The introduction of IP-based real-time services in next-generation mobile systems requires coupling mobility with quality of service. The mobility of the node can disrupt or even intermittently disconnect an ongoing real-time session. The duration of such an interruption is called disruption time or handover latency, and can heavily affect user satisfaction. Therefore, this delay needs to be minimized to provide good quality of VoIP services. In this article, we focus on network-layer mobility and mobile IP since it is a natural candidate for providing such mobility. We evaluate different low-latency schemes based on mobile IP and compare their performances in terms of disruption time for VoIP services. Low-latency handoffs are performed by anticipating and/or postponing the mobile IP registration process. With these methods, disruption time is reduced to 200 ms in most considered cases.     

Network‐initiated simultaneous mobility in voice over 3GPP‐WLAN

VoIP over WLAN (VoWLAN) gradually has become a popular application with the fast maturing of both WLAN and Voice over IP (VoIP) technology. However there exists one problem that heavily affects the satisfaction of the users which is that the mobility of the mobile host (MH) can disrupt or even intermittently disconnect an ongoing real‐time session. Therefore the issue of how to reduce the handover delay gets more and more important. This paper proposes a Network‐Initiated SimUltaneouS mobility (NISUS) mechanism to facilitate terminal mobility with the session initiation protocol (SIP) in Voice over 3GPP‐WLAN. We design the E2E tunnel state model running on the packet data gateway (PDG) referring to the CAMEL concept. The NISUS is triggered at the PDG by detecting the state transition of the E2E tunnel state model that represents the occurrence of a handover. Then the PDG sends the handover request to notify the Mobility Server (MS) to perform a third party call control (3PCC) and a third party registration on behalf of the MH in parallel for session re‐establishment. With the help of the MS we ensure the lost signaling messages could be correctly re‐sent to moving hosts. Moreover the Master‐Slave Determination procedures derived from H.245 are proposed for the MS in order to handle the racing conditions fairly when two MSs involved in a simultaneous mobility issue 3PCC calls respectively at about the same time. We demonstrate the NISUS works well in the simultaneous and non‐simultaneous movement cases. Analytical results show that the handover delay can be improved significantly by using the NISUS compared with the mobile‐initiated simultaneous/non‐simultaneous mobility. Copyright © 2010 John Wiley & Sons, Ltd.     

An Approach to Integrating SIP in Converged Multimodal/Multimedia Communication Services

In this paper, we present an approach of integrating SIP (Session Initiation Protocol) in converged multimodal/multimedia communication services. An extensible VoIPTeleserver for VoIP in SIP environment is described. It is based on the concept of dialogue system and Web convergence that separates the channel dependent media resources from the application dependent service creation and hosting environment. It supports XML based service applications for multiple channels including voice, DTMF, IM and chat over IP. The loosely coupled open architecture in our approach is highly extensible. We describe the concept and structure of VoIPTeleServer used in our approach in detail, which interfaces to the VoIP world through SIP signaling and works as a broker between the VoIP SIP environment and MTIP to deliver converged communication services. A prototype of VoIPTeleServer was implemented, and services and applications based on SIP and MTIP convergence are constructed. Special attention is given to the adverse effect of delay, jitter and packet loss for voice portal services over IP. In particular, case studies of DTMF service in voice portal under adverse channel conditions are performed. The compounding effects of multiple channel impairments to DTMF in voice portal services over IP are characterized. The potential high error rate of the DTMF service indicates that the data redundancy method as proposed in RFC 2198 is needed for DTMF in order to achieve reliable voice portal services over IP.     

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。     

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。     

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).     

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.     

现代通信技术与VoIP

分析了公用交换电话网络工作原理,分析了在因特网和基于IP的数据网上传送语音业务技术的协议标准及其在市场应用方面的优势,并给出了一个VoIP组网模型。     

基于Cable Modem技术的VoIP话音解决方案

目前,一些大、中城市的有线电视公司已经拥有庞大的有线电视宽带用户群,而基于HFC网的VoIP系统正是目前在有线电视宽带网条件下综合接入语音、数据和多媒体业务的最佳解决方案之一,其中语音业务可利用中继媒体网关设备完成媒体流转换,并通过标准的E1协议接入PSTN网络,以达到充分利用HFC及PSTN网络资源迅速开展话音业务的目的.同时,作为NGN(下一代网络)中的标准部件,VoIP是面向未来、可持续发展的解决方案之一,在有线电视宽带网所及之处,可以为商业和家庭用户提供质优价廉的IP语音服务.     

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     

可堆叠VoIP语音板卡的固件程序设计

本文提出了基于口可堆叠式的VoIP通信应用系统的系统架构,重点介绍了面向IP可堆叠式的VoIP语音板卡的固件程序设计.每块VoIP语音板卡支持8路语音,通过自定义的通信协议可使不同的VoIP语音板卡独立地通过IP互联,实现基于IP可堆叠.自定义通信协议实现了VoIP语音板卡中芯片内部通道之间、VoIP语音板卡上芯片之间、不同VoIP语音板卡之间,以及VoIP语音板卡与管理PC间的通信.VoIP语音板卡控制软件以内核模块方式运行,并在内核模块方式下由VINETIC-2CPE语音芯片中断服务程序激活回调函数,提高了实时性.     

Designing VoIP session management over interworked WLAN-3G networks

In an all-IP internetworked heterogeneous environment, ongoing VoIP sessions from roaming users will be subject to frequent vertical handoffs across network boundaries. Ensuring uninterrupted service continuity for these handoff calls requires successful session management among the participating access networks. As such, a mobility-aware novel interworking network design (interconnecting UMTS and WLAN over an IP-based common platform) [1] is presented in this article that facilitates VoIP session management, including session establishment and seamless session handoff across different networks. For comparison purposes, VoIP session management is evaluated in terms of session establishment, handoff delays, transient packet loss, end-to-end traffic delays, and jitter value for different voice codecs, which demonstrate satisfactory and feasible results. In the event (e.g., network congestion, buffer overflow) that session continuity cannot be guaranteed (also known as outage) across network boundaries, this article proposes an algorithm that compensates the user by reducing the unit service charge of future sessions (governed by the outage period) through a noncooperative game-theory-based pricing mechanism.     

An energy‐efficient transmission mechanism for VoIP over IEEE 802.11 WLAN

Voice over IP (VoIP) over WLAN (VoWLAN) is an important application for public and private WLANs. However, VoWLAN systems suffer from several technical challenges such as power consumption of a WLAN station (STA) and service capacity of an access point (AP), making the commercial deployment of a large‐scale VoWLAN service problematic. This study presents a cross‐layer and energy‐efficient mechanism for transmitting VoIP packets over IEEE 802.11 WLAN. The proposed mechanism considers the characteristics of voice packets that can tolerate certain loss, and dynamically disables the medium access control (MAC) layer acknowledgement for voice packets. In doing so, the time and energy consumed to transmit and receive voice packets for an STA can be reduced. Simulation results demonstrate that the mechanism improves the energy efficiency of a VoWLAN STA and WLAN utilization without sacrificing voice qualities. Copyright © 2009 John Wiley & Sons, Ltd.     

VoIP — Panacea or PIG's Ear?

Voice over IP (VoIP) hit the headlines during the mid-1990s amid claims concerning its potential impact upon existing switched-circuit telephony services. While VoIP has provided a focus for much debate within the industry, there has been a clear gulf between the marketing hype and the technological reality of what VoIP really is and what it can offer. This paper examines VoIP as a technology, considering some of the drivers for meriting its application within the communications industry, introduces the key aspects that need to be considered and indicates the nature of the scope of opportunities afforded.