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.     

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     

下一代网络的业务互通体系结构

以IP为基础的因特网的出现对传统的电信网络带来了巨大的冲击,它的开放性和业务的灵活性为电信网进一步的发展指出了一个方向.但是,由于IP在提供具有质量保证的等级业务方面缺乏技术的保障;局域网技术、广域网技术的发展使得原来被认为功能简单的第二层传输网络具有越来越强的业务能力,例如QoS、多播、VPN以及安全管理能力,将导致多接入、多业务的下一代网络端到端业务控制和业务集成越来越复杂,使得我们需要重新审视网络互连和业务集成的基本思路,建立下一代网络的新的业务概念.本文对这个问题进行了初步的探讨,结论是:下一代网络具有新的集成业务体系结构,其中的关键是业务互通问题;其中IP用于网络互联,SIP用于建立终端和服务器间的业务会话,提供业务之间的互作用,实现端到端业务能力的集成和最终应用的集成需要新的网格业务技术;下一代网络除要有分别支持网络互联、会话建立和业务集成的能力以外,还必须建立网络、会话和业务之间的新的互作用关系,使得下一代网络的目标得以实现.     

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.     

COPS-SLS usage for dynamic policy-based QoS management over heterogeneous IP networks

While each IP domain can deploy its own strategy to manage network resources, multimedia traffic needs end-to-end QoS management to obtain an overall service level. The provision of end-to-end QoS over a heterogeneous environment implies the negotiation of a mutually acceptable SLA. This article presents the use of the COPS-SLS protocol as a generic protocol for automatic service-level negotiation and the integration of this protocol in an overall QoS management architecture to manage service levels over multiple domains deploying different QoS technologies.     

A new call control scheme of Push-To-Talk system

For Push-To-Talk (PTT) system based on Public Mobile Data Network (PMDN), the end-to-end time delay is the key aspect of the user’s experience. The Push-Over-Cellular (POC) scheme defined by Open Mobile Alliance (OMA) is based on the VoIP phone model and use SIP protocol as the call control scheme. The call setup time delay in SIP may reach to several seconds, which is unacceptable for the PTT service. In this paper, we provide a new call control scheme for PTT system based on PLMN network. By combining the apriority knowledge of PTT call model and the priority control scheme, we encapsulate the signaling message and the voice data into a same data packet, when the user push the button, the voice and the call control signaling are sent to the server at the same time. So the long time delay of call setup procedure of POC scheme can be eliminate. The end-to-end call delay can be decreased significantly. The experiment result based on the commercial CDMA2000 1X network of China Unicom shows that the call delay can be decreased to 600 ms, which approach to the traditional trunk communication system’s requirement.     

IP电话发展新动向

介绍了NTT公司为参与到IP电话领域而进行的可行性论证、公司内部IP电话实验以及面向商业用户而推出的三种VoIP服务。对新的IP电话网从呼叫控制、通话建立、QoS和编译码、计费方面进行了介绍，并对IP国际电话、呼叫中心及新推出的服务进行了说明。     

Resource management for QoS support in cellular/WLAN interworking

To provide mobile users with seamless Internet access anywhere and anytime/ there is a strong demand for interworking mechanisms between cellular networks and wireless local area networks in the next-generation all-IP wireless networks. In this article we focus on resource management and call admission control for QoS support in cellular/WLAN interworking. In specific, a DiffServ interworking architecture with loose coupling is presented. Resource allocation in the interworking environment is investigated/ taking into account the network characteristics, vertical handoff, user mobility, and service types. An effective call admission control strategy with service differentiation is proposed for QoS provisioning and efficient resource utilization. Numerical results demonstrate the effectiveness of the proposed call admission control scheme.     

The convergent phone

Past claims for Internet telephony have proved over optimistic, but the technology remains crucial to the future of the telecoms business. The author reports on how the industry is learning to live with the challenge of network convergence. The development effort required to deliver appropriate QoS levels and seamless interworking with the PSTN has been massive, and is ongoing. Within the closed world of single-company autonomous networks, or the confines of a company WAN/LAN, the technology is now practicable, but it's an open question as to whether VoIP will ever operate across multiple networks with the performance and reliability of the PSTN. Such reservations aside, the operational and service attractions of VoIP will make the successful management of convergence the most important challenge for network operators over the coming decade     

QoS issues in the converged 3G wireless and wired networks

The Internet evolution delineated through the last years has urged the wireless network community to support the deployment of IP multimedia services with guaranteed quality of service (QoS) in 3G wireless networks. This article copes with the interoperability between 3G wireless networks and wired next-generation IP networks, for the provision of services with an a priori known quality level over both environments. More specifically, the UMTS architecture as well as a prototypical implementation of the next-generation Internet based on DiffServ are considered. The article focuses on the mapping among the traffic classes of the two networks at the point where the networks converge, and discusses the requirements and possible solutions for their proper interworking at the signaling and user levels. Simulations prove that proper mapping among the traffic classes of each world is necessary in order to achieve the desired end-to-end traffic characteristics.     

QoS and service interworking using constraint route labeldistribution protocol (CR-LDP)

Introducing quality of service features to the IP/TCP protocol suite has become a hot topic of research in both industry and academia. Several architectures have been proposed for QoS support at the network layer (layer 3 in the OSI model). Both integrated services and differentiated services architectures are examples of QoS models that are implementable at the IP layer. Another development at the IETF is the work related to traffic engineering using multiprotocol label switching. While traffic engineering covers a wide range of topics, QoS support is recognized as one of its necessary features. This article describes the QoS features of the constraint-route label distribution protocol and how they can be efficiently utilized to achieve service interworking between a number of different networking technologies     

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.     

Toward an all-IP-based UMTS system architecture

Looking into the future, two main drivers for the mobile telecommunications market can be identified: third-generation mobile systems (e.g., UMTS) and the Internet (e.g., the introduction of IP technologies like voice/multimedia over IP in mobile networks). UMTS is seen as the enabler of wireless multimedia applications and portability of a personalized service set across network/terminal boundaries, as defined within the virtual home environment (VHE) system concept. In light of these evolutions, this article investigates the impact of the evolution toward an all-IP UMTS network architecture on the UMTS service architecture, which is based on the VHE concept. The article discusses two possible scenarios for supporting VoIP services in the UMTS service architecture and analyzes their applicability in an all-IP-based UMTS network. The first is based on the traditional centralized IN service architecture. The second proposes a new decentralized architecture based on direct control of VoIP call control equipment by open service architecture interfaces     

VoIP基础电信服务的运营环境探讨

从普遍服务、互联互通、用户名/编号、执法监听、紧急呼叫与基于位置的监管等六方面介绍了政府管制的内容,并分析了其对VoIP运营环境的影响.阐明了VoIP运营环境中可能的资费组成具有的特征：VoIP业务使用与承载网络资源使用分别计费;同一运营商内部实行低廉的VoIP基础资费;不同运营商之间的业务互通支付单独费用;服务质量和安全保障将成为附加的VoIP服务.     

A differentiated optical services model for WDM networks

This article addresses the issues of scalable end-to-end QoS in metropolitan DWDM networks serving as transit networks for IP access networks. DWDM offering a few wavelengths has been deployed in the past in backbone networks to upgrade point-to-point transmission where sharing is based on coarse granularity. This type of DWDM backbone network, offering a few light-paths, provides no support for QoS services traversing the network. As DWDM networks with larger numbers of wavelengths penetrate the data-centric metro environment, specific IP service requirements such as priority restoration, scalability, dynamic provisioning of capacity and routes, and support for coarse-grain QoS capabilities will have to be addressed in the optical domain in order to achieve end-to-end QoS over a DWDM network. We propose a QoS service model in the optical domain called differentiated optical services (DoS) based on a set of optical parameters that captures the quality and reliability of the optical lightpath     

Toward scalable admission control for VoIP networks

We present an overview of scalable admission control in IP networks. We introduce various approaches and discuss the mechanism and characteristics of each method. In particular, we argue that end-to-end measurement based admission control (EMBAC), which employs end-to-end on-demand probing, should be used for call admission control. Second, we consider use of EMBAC in VoIP networks. We present a new probability-based EMBAC scheme and show that its performance is close to the ideal method using virtual-trunk-based admission control. We also present a QoS allocation approach for selecting an admission threshold and dimensioning link capacities. A simple network design and evaluation results suggest that this QoS allocation approach is effective to adequately dimension a network, while satisfying end-to-end targets in terms of blocking probability and packet loss rate.     

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语音时延的分析和研究

文章介绍了VoIP(IP网络上传送语音)语音质量的测试方法,分析了影响VoIP语音质量的主要因素:延迟、抖动、丢包率和时延.利用E模型定量地分析了语音质量与端到端时延的关系,通过建立数学模型,指出了VoIP 系统中主要的时延分量,并研究了这些时延分量产生的机理和影响它们的参数.在设计实际的VoIP系统时,可以通过优化影响时延分量的主要参数,改善VoIP系统的时延.     

Transparent IP radio access for next-generation mobile networks

Advances in network architecture, enhancements in signaling protocols, provisioning of end-to-end QoS, worldwide seamless mobility, and flexible service provision are among the major research challenges toward next-generation wireless networks. The integration and interoperability of all these technologies, along with new truly broadband wireless innovations and intelligent user-oriented services will lead toward the so-called 4G wireless networks. In this article we identify the key issues of an innovative transparent IP radio access system that targets 4G networks.     

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.