Efficient Architecture and Handoff Strategy used for VoIP Sessions in SIP Based Wireless Networks

In the near future, the Internet is likely to become an All-IP network that provides various multimedia services over wireless networks. Although the earliest VoIP applications did not consider the end-node mobility, researchers have attempted to support mobility in current VoIP protocols, such as Session Initial Protocol (SIP)-based mobility. The SIP-based mobility is considered because it can readily support mobility. However, calling disruptions may occur in traditional SIP mid-call terminal mobility because handoff procedure may be required, depending on the implementation and the real network deployment considerations. In any case, issues in the combined SIP/RSVP for guaranteeing QoS of VoIP service under mobile environment are also considered to be crucial. Therefore, this study describes the solutions by devising novel hierarchy network architecture. Also, the mechanisms including help with neighboring users in adjacent cells and the third party call control to overcome those issues are included. The simulation results indicate that the proposed technique is practical and better executive than conventional schemes.     

Performance of delay‐sensitive traffic in multi‐layered satellite IP networks with on‐board processing capability

In this article, performance of delay‐sensitive traffic in multi‐layered satellite Internet Protocol (IP) networks with on‐board processing (OBP) capability is investigated. With OBP, a satellite can process the received data, and according to the nature of application, it can decide on the transmission properties. First, we present a concise overview of relevant aspects of satellite networks to delay‐sensitive traffic and routing. Then, in order to improve the system performance for delay‐sensitive traffic, specifically Voice over Internet Protocol (VoIP), a novel adaptive routing mechanism in two‐layered satellite network considering the network's real‐time information is introduced and evaluated. Adaptive Routing Protocol for Quality of Service (ARPQ) utilizes OBP and avoids congestion by distributing traffic load between medium‐Earth orbit and low‐Earth orbit layers. We utilize a prioritized queueing policy to satisfy quality‐of‐service (QoS) requirements of delay‐sensitive applications while evading non‐real‐time traffic suffer low performance level. The simulation results verify that multi‐layered satellite networks with OBP capabilities and QoS mechanisms are essential for feasibility of packet‐based high‐quality delay‐sensitive services which are expected to be the vital components of next‐generation communications networks. Copyright © 2007 John Wiley & Sons, Ltd.     

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     

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.     

基于软交换体系下的VoIP技术的实现

VoIP技术是下一代网络发展的新技术。论文介绍了VoIP在NGN网络中的演变,提山了基于软交换体系下的VoIP结构,并且根据软交换能够使各协议互通的特点,尽量使各协议在VoIP结构中发挥各自的优点,使其在通话质量、维护管理达到最优化,并能容易地增加另外的增值服务。     

Quality assurance of voice over WLANs (VoWLANs) with differentiated services

Several technical issues make commercial and large voice over wireless local area network (VoWLAN) services difficult to provide. The most challenging issue when voice over Internet Protocol (VoIP) services are ran over IEEE 802.11-based WLANs is the bandwidth inefficiency due to the considerable overhead associated with WLAN packet transmission. In this work, we propose a session-based quality-of-service management architecture (SQoSMA) to overcome the low number of VoIP calls in IEEE 802.11 Wireless LANs and the negative effect of new call addition when the WLAN reaches its capacity. The SQoSMA combines data and control planes to detect VoWLAN QoS degradations and performs either an adaptive audio codec switching or a call stopping to fix VoWLAN issues in a differentiated services manner. In addition, our solution deals with user sessions information, by considering user priority (from its agreement) to guarantee a certain level of its multimedia applications. Performance evaluation using a real test-bed shows that call codec change and call stopping techniques can easily assure high-priority calls with acceptable call blocking probability.     

Providing statistical QoS guarantee for voice over IP in the IEEE 802.11 wireless LANs

Recent years have seen greatly increasing interests in voice over IP in wireless LANs, in which the IEEE 802.11 distributed coordination function protocol or enhanced DCF protocol is used. However, since both DCF and EDCF are contention-based medium access control protocols, it is difficult for them to support the strict QoS requirement for VoIP. Therefore, in this article we propose a novel call admission control scheme that runs at the MAC layer to support VoIP services. The call admission control mechanism regulates voice traffic to efficiently coordinate medium contention among voice sources. The rate control mechanism regulates non-voice traffic to control its impact on the performance of voice traffic. Extensive simulations demonstrate that the proposed schemes can well support statistical QoS guarantees for voice traffic and maintain stable high throughput for non-voice traffic at the same time.     

DiffServ resource allocation for fast handoff in wireless mobileInternet

The next-generation wireless networks are evolving toward a versatile IP-based network that can provide various real-time multimedia services to mobile users. Two major challenges in establishing such a wireless mobile Internet are support of fast handoff and provision of quality of service (QoS) over IP-based wireless access networks. In this article, a DiffServ resource allocation architecture is proposed for the evolving wireless mobile Internet. The registration-domain-based scheme supports fast handoff by significantly reducing mobility management signaling. The registration domain is integrated with the DiffServ mechanism and provisions QoS guarantee for each service class by domain-based admission control. Furthermore, an adaptive assured service is presented for the stream class of traffic, where resource allocation is adjusted according to the network condition in order to minimize handoff call dropping and new call blocking probabilities     

Low‐cost and high‐performance: VoIP monitoring and full‐data retention at multi‐Gb/s rates using commodity hardware

Voice over IP (VoIP) is increasingly replacing the old public switched telephone network (PSTN) technology. In this new scenario, there are several challenges for VoIP providers. First, VoIP requires a detailed monitoring of both users' quality of service (QoS) and experience (QoE) to a greater extent than in traditional PSTNs. Second, such a monitoring process must be able to track VoIP traffic in high‐speed networks, nowadays typically of multi‐Gb/s rates. Third, recent government directives require that providers retain information from their users' calls. Similarly, the convergence of data and voice services allows operators to provide new services such as full‐data retention, in which users' calls can be recorded for either quality assessment (call centers, QoE) or security purposes (lawful interception). This implies a significant investment in infrastructure, especially in large‐scale networks which require multiple points of measurement and redundancy. This paper proposes a novel methodology, architecture and system to fulfill such challenges, called VoIPCallMon, as well as the data structures and necessary hardware‐tuning knowledge for its development. As distinguishing features, VoIPCallMon provides very high performance, being able to process VoIP traffic on‐the‐fly at high bitrates, novel services and significant cost reduction by using commodity hardware with minimal interference with operational VoIP networks. The performance evaluation shows that the system copes with the VoIP load of real‐world operators. We further evaluated the system performance at a fully saturated 10 Gb/s link and no packet loss was reported, therefore demonstrating the potential of commodity hardware solutions. Copyright © 2014 John Wiley & Sons, Ltd.     

互联网骨干网流量的建模和特征

With enormous growth of the number of Internet users and appearance of new applications, characterization of Internet traffic has attracted more and more attention and has become one of the major challenging issues in telecommunication network over the past few years. In this paper, we study the network traffic pattern of the aggregate traffic and of specific application traffic, especially the popular applications such as P2P, VoIP that contribute most network traffic. Our study verified that majority Internet backbone traffic is contributed by a small portion of users and a power function can be used to approximate the contribution of each user to the overall traffic. We show that P2P applications are the dominant traffic contributor in current Internet Backbone of China. In addition, we selectively present the traffic pattern of different applications in detail.     

VoIP traffic in wireless mesh networks: a MOS‐based routing scheme

Support of Voice over Internet Protocol (VoIP) services in wireless mesh networks requires implementation of efficient policies to support low‐delay data delivery. Multipath routing is typically supported in wireless mesh networks at the network level to provide high fault tolerance and load balancing because links in the proximity of the wireless mesh gateways can be very stressed and overloaded, thus causing scarce performance. As a consequence of using multipath solutions, lower delay and higher throughput can be supported also when a given path is broken because of mobility or bad channel conditions, and alternative routes are available. This can be a relevant improvement especially when assuming that real‐time traffic, such as VoIP, travels into the network. In this paper, we address the problem of Quality of Service (QoS) support in wireless mesh networks and propose a multipath routing strategy that exploits the Mean Opinion Score (MOS) metric to select the most suitable paths for supporting VoIP applications and performing adaptive load balancing among the available paths to equalize network traffic. Performance results assess the effectiveness of the proposed approach when compared with other existing methodologies. Copyright © 2015 John Wiley & Sons, Ltd.     

100+ VoIP Calls on 802.11b: The Power of Combining Voice Frame Aggregation and Uplink-Downlink Bandwidth Control in Wireless LANs

The bandwidth efficiency of voice over IP (VoIP) traffic on the IEEE 802.11 WLAN is notoriously low. VoIP over 802.11 incurs high bandwidth cost for voice frame packetization and MAC/PHY framing, which is aggravated by channel access overhead. For instance, 10 calls with the G.729 codec can barely be supported on 802.11b with acceptable QoS - less than 2% efficiency. As WLANs and VoIP services become increasingly widespread, this inefficiency must be overcome. This paper proposes a solution that boosts the efficiency high enough to support a significantly larger number of calls than existing schemes, with fair call quality. The solution comes in two parts: adaptive frame aggregation and uplink/downlink bandwidth equalization. The former reduces the absolute number of MAC frames according to the link congestion level, and the latter balances the bandwidth usage between the access point (AP) and wireless stations. When used in combination, they yield superior performance, for instance, supporting more than 100 VoIP calls over an IEEE 802.11b link. The authors demonstrate the performance of the proposed approach through extensive simulation, and validate the simulation through analysis.     

Assessing the quality of voice communications over Internet backbones

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.     

Service overlay networks: SLAs, QoS, and bandwidth provisioning

We advocate the notion of service overlay network (SON) as an effective means to address some of the issues, in particular, end-to-end quality of service (QoS), plaguing the current Internet, and to facilitate the creation and deployment of value-added Internet services such as VoIP, Video-on-Demand, and other emerging QoS-sensitive services. The SON purchases bandwidth with certain QoS guarantees from the individual network domains via bilateral service level agreement (SLA) to build a logical end-to-end service delivery infrastructure on top of the existing data transport networks. Via a service contract, users directly pay the SON for using the value-added services provided by the SON. In this paper, we study the bandwidth provisioning problem for a SON which buys bandwidth from the underlying network domains to provide end-to-end value-added QoS sensitive services such as VoIP and Video-on-Demand. A key problem in the SON deployment is the problem of bandwidth provisioning, which is critical to cost recovery in deploying and operating the value-added services over the SON. The paper is devoted to the study of this problem. We formulate the bandwidth provisioning problem mathematically, taking various factors such as SLA, service QoS, traffic demand distributions, and bandwidth costs. Analytical models and approximate solutions are developed for both static and dynamic bandwidth provisioning. Numerical studies are also performed to illustrate the properties of the proposed solutions and demonstrate the effect of traffic demand distributions and bandwidth costs on SON bandwidth provisioning.     

Performance Optimizations for Deploying VoIP Services in Mesh Networks

In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients     

iVoIP: an intelligent bandwidth management scheme for VoIP in WLANs

Voice over Internet Protocol (VoIP) has been widely used by many mobile consumer devices in IEEE 802.11 wireless local area networks (WLAN) due to its low cost and convenience. However, delays of all VoIP flows dramatically increase when network capacity is approached. Additionally, unfair traffic distribution between downlink and uplink flows in WLANs impacts the perceived VoIP quality. This paper proposes an intelligent bandwidth management scheme for VoIP services (iVoIP) that improves bandwidth utilization and provides fair downlink–uplink channel access. iVoIP is a cross-layer solution which includes two components: (1) iVoIP-Admission Control, which protects the quality of existing flows and increases the utilization of wireless network resources; (2) iVoIP-Fairness scheme, which balances the channel access opportunity between access point (AP) and wireless stations. iVoIP-Admission Control limits the number of VoIP flows based on an estimation of VoIP capacity. iVoIP-Fairness implements a contention window adaptation scheme at AP which uses stereotypes and considers several major quality of service parameters to balance the network access of downlink and uplink flows, respectively. Extensive simulations and real tests have been performed, demonstrating that iVoIP has both very good VoIP capacity estimation and admission control results. Additionally, iVoIP improves the downlink/uplink fairness level in terms of throughput, delay, loss, and VoIP quality.     

An Adaptive QoS Power Saving Scheme for Mobile WiMAX

With the rapid development of Internet and wireless communication technology, ubiquitous network services become more and more popular. WiMAX is widely used to solve the last mile in network deployment. To enhance the mobility, mobile WiMAX is launched to support the mobile usages. Due to the limited power of mobile devices, power saving becomes a key issue for mobile WiMAX applications. Though plenty of efforts have been proposed to save power on the mobile devices, the problems are partially solved. Among others, sleep mode operations are widely adopted to save power in wireless communications. Generally, longer sleep time can reduce power consumption at the cost of increased packet response delay. To improve the quality of services, an adaptive power saving scheme for mobile WiMAX is proposed in this paper. Parameters related to power management are dynamically set according to current network traffic load. The analyses and simulation results show that the proposed scheme presents superior power efficiency and packet response delay in the context of mobile WiMAX.     

Understanding and improving TCP performance over networks withminimum rate guarantees

A large number of Internet applications are sensitive to overload conditions in the network. While these applications have been designed to adapt somewhat to the varying conditions in the Internet, they can benefit greatly from an increased level of predictability in network services. We propose minor extensions to the packet queueing and discard mechanisms used in routers, coupled with simple control mechanisms at the source that enable the network to guarantee minimal levels of throughput to different sessions while sharing the residual network capacity in a cooperative manner. The service realized by the proposed mechanisms is an interpretation of the controlled-load service being standardized by the Internet Engineering Task Force. Although controlled-load service can be used in conjunction with any transport protocol, our focus in this paper is on understanding its interaction with Transmission Control Protocol (TCP). Specifically, we study the dynamics of TCP traffic in an integrated services network that simultaneously supports both best-effort and controlled-load sessions. In light of this study, we propose and experiment with modifications to TCP's congestion control mechanisms in order to improve its performance in networks where a minimum transmission rate is guaranteed. We then investigate the effect of network transients, such as changes in traffic load and in service levels, on the performance of controlled-load as well as best-effort connections. To capture the evolution of integrated services in the Internet, we also consider situations where only a selective set of routers are capable of providing service differentiation between best-effort and controlled-load traffic. Finally, we show how the service mechanisms proposed here can be embedded within other packet and link scheduling frameworks in a fully evolved integrated services Internet     

WLAN中基于MIPv6的移动VoIP切换性能仿真分析

移动IP是一个在Internet上基于网络层提供移动性支持功能的要求较高的VoIP业务,切换延迟将直接影响到话音质量,严重时甚至会中断正在进行的会话.文章借助ns2网络模拟器仿真分析了WLAN中基于MIPv6的移动VoIP切换性能.结果表明,MIPv6及其扩展协议的切换性能优劣顺序依次为:F-HMIPv6、FMIPv6、HMIPv6、MIPv6.尤其是F-HMIPv6协议,无论端到端延迟还是切换延迟,都得到了最大的改善.所得结论能为网络切换性能的进一步优化提供重要依据.