An analytical simulator for deploying IP telephony

Deploying IP telephony or voice over IP (VoIP) is a major and challenging task. This paper describes an analytical design and planning simulator to assess the readiness of existing IP networks for the deployment of VoIP. The analytical simulator utilizes techniques used for network flows and queuing network analysis to compute two key performance bounds for VoIP: delay and bandwidth. The simulator is GUI‐based and has an interface with drag‐and‐drop features to easily construct any generic network topology. The simulator has an engine that automates and implements the analytical techniques. The engine determines the number of VoIP calls that can be sustained by the constructed network while satisfying VoIP QoS requirements and leaving adequate capacity for future growth. As a case study, the paper illustrates how the simulator can be utilized to assess the readiness to deploy VoIP for a typical network of a small enterprise. We have made the analytical simulator publicly available in order to improve and ease the process of VoIP deployment. Copyright © 2008 John Wiley & Sons, Ltd.     

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.     

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.     

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.     

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.     

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.     

一种动态时分窄带多业务接入新方案

如何在窄带低比特率链路上进行高效的语音数据等多业务综合接入,并保证语音等实时业务的质量是目前多业务接入的一个重点问题.本文提出的动态时分多业务接入方案解决了Digital Data Network (DDN)专线上窄带压缩语音,ADPCM语音,传真,以及数据的同时接入问题,有效地解决了DDN专线上多业务接入的质量保证问题,提高了链路利用率.本文对于该方案的性能进行了分析,并与目前基于IP的多业务接入方案进行了比较.本方案能够提供目前的VoIP方案下所无法提供的语音,传真业务的质量保证,在多业务的支持上比VoIP更加简单,更有吸引力.     

Contention‐free approaches for WiFi MAC design for VoIP services: performance analysis and comparison

The exponential growth in the demand of voice over internet protocol (VoIP) services along with the increasing demand for mobility in VoIP services has attracted great research efforts towards provisioning of VoIP services in IEEE 802.11‐based Wireless LANs (WiFi networks). We address one of the important research problems, namely, the quality of service (QoS)‐aware efficient silence suppression in the bursty voice traffic, for provisioning VoIP services in WiFi networks. The research works in the recent literature on silence suppression in voice calls have been surveyed categorising them on how the activity arrival is notified to the access point (AP). In most of the recent schemes, notification of uplink activity arrival is done through contention based medium access mechanisms such as the distributed coordination function (DCF). Contention‐based medium access causes non‐deterministic delays, therefore such schemes are not suited to voice traffic which require strict delay bound guarantees. This paper focuses on the schemes which do not use contention based approaches for silence suppression in voice traffic. Analytical performance evaluation and comparison of such schemes is carried out. Two very important performance metrics are modelled mathematically. One is the expected polling overhead time that the schedulers in these schemes can save per voice call during one voice activity cycle as compared to that in the round‐robin polling scheduler. The other is the expected unnecessary wireless channel access delay that a typical first talk‐spurt frame experiences due to the specific design of each scheme. The numerical results of this evaluation lead us to the conclusion whether or not and to what extent each of these schemes is viable. Copyright © 2008 John Wiley & Sons, Ltd.     

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.     

面向固定移动融合环境的P2PSIP系统设计

全业务运营是电信市场继语音和宽带接入服务之后的下一个增长点,而基于IP的融合有线网络和无线网络的语音服务则是全业务的重点之一。本文通过分析现有VoIP网络存在的问题以及固定移动融合网络环境下VoIP的特点,提出一种新型双层重叠网架构的P2PSIP架构,并阐述了新型架构的优点及双层重叠网之间的通信机制。新型架构能有效提高系统的安全性、健壮性和用户节点资源利用效率,更好的满足固定移动融合网络环境下VoIP对带宽、网络质量和安全性的要求。     

QoS enabled voice support in the next generation Internet: issues,existing approaches and challenges

The Internet is under rapid growth and continuous evolution in order to accommodate an increasingly large number of applications with diverse service requirements. In particular, Internet telephony, or voice over IP is one of the most promising services currently being deployed. Besides the potentially significant cost reduction, Internet telephony can offer many new features and easier integration with widely adopted Web-based services. Despite these advantages, there still exist a number of barriers to the widespread deployment of Internet telephony. The most prominent one, however, is how to ensure the QoS needed for voice conversation. The purpose of this article is to survey the state-of-the-art technologies in enabling the QoS support for voice communications in the next-generation Internet. In this article, we first review the existing technologies in supporting voice over IP networks, including the basic mechanisms in the IETF Internet telephony architecture and ITU-T H.323-related Recommendations. We then discuss the IETF QoS framework, specifically the Intserv and Diffserv framework. Finally, we present two leading companies' (Cisco and Lucent) solutions to offering IP telephony services as examples to illustrate how real systems are implemented     

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.     

Type-2 fuzzy logic control of PQoS driven adaptive VoIP scheme

Adaptive VoIP schemes have potentially suboptimal performance owing to imprecision in the metrics used to infer network state. An interval Type-2 fuzzy logic controlled scheme for VoIP services is presented. It infers network state from average delivered perceived quality of service and its degradation due to network congestion and updates an AMR codec mode to match voice quality to available network bandwidth. Tests showed that the scheme maximised delivered voice quality and outperformed an existing adaptive scheme. The scheme achieves robust performance in the presence of input imprecision and can be implemented in VoIP terminals, and the fuzzy rule base is easy to understand and change by non-experts because of its similarity to the human decision-making process.     

NCTU‐VT: a freeware for wireless VoIP performance measurement

Voice over IP (VoIP) is a promising low‐cost voice communication over the wireless IP network. To provide satisfactory VoIP services, the Quality of Service (QoS) of the wireless network should be guaranteed. This paper proposes a VoIP performance measurement freeware called NCTU VoIP Testing Tool (NCTU‐VT). We compare NCTU‐VT with two commercial tools SmartVoIPQoS and IxChariot in terms of packet loss, latency, and Mean Opinion Score (MOS) of the VoIP sessions in Wi‐Fi network. Our study indicates that these three tools can accurately measure VoIP performance in Wi‐Fi environment. Copyright © 2010 John Wiley & Sons, Ltd.     

The extended connection‐dependent threshold model for call‐level performance analysis of multi‐rate loss systems under the bandwidth reservation policy

Firstly, we reviewed two extensions of the Erlang multi‐rate loss model, whereby we can assess the call‐level QoS of telecom networks supporting elastic traffic: (i) the extended Erlang multi‐rate loss model, where random arriving calls of certain bandwidth requirements at call setup can tolerate bandwidth compression while in service; and (ii) the connection‐dependent threshold model, where arriving calls may have several contingency bandwidth requirements, whereas in‐service calls cannot tolerate bandwidth compression. Secondly, we proposed a new model, the extended connection‐dependent threshold model. Calls may have alternative bandwidth requirements at call setup and can tolerate bandwidth compression while in service. We proposed a recurrent formula for the efficient calculation of link occupancy distribution and consequently call blocking probabilities, link utilization, and throughput per service class. Furthermore, in the proposed model, we incorporated the bandwidth reservation policy, whereby we can (i) equalize the call blocking probabilities of different service classes, (ii) guarantee specific QoS per service class, and (iii) implement different maximum bandwidth compression/expansion rate per service class so that the network supports both elastic and stream traffic. The accuracy of the new model is verified by simulation. Moreover, the proposed model performs better than the existing models. Finally, we generalize the proposed model by incorporating service classes with either random or quasi‐random arrivals. Copyright © 2011 John Wiley & Sons, Ltd.     

Radio Resource Allocation for Scalable Video Services Over Wireless Cellular Networks

Good quality video services always require higher bandwidth. Hence, to provide the video services e.g., multicast/broadcast services (MBSs) and unicast services along with the existing voice, internet, and other background traffic services over the wireless cellular networks, it is required to efficiently manage the wireless resources in order to reduce the overall forced call termination probability, to maximize the overall service quality, and to maximize the revenue. Fixed bandwidth allocation for the MBS sessions either reduces the quality of the MBS videos and bandwidth utilization or increases the overall forced call termination probability and of course the handover call dropping probability as well. Scalable video coding (SVC) technique allows the variable bit rate allocation for the video services. In this paper, we propose a bandwidth allocation scheme that efficiently allocates bandwidth among the MBS sessions and the non-MBS traffic calls (e.g., voice, unicast, internet, and other background traffic). The proposed scheme reduces the bandwidth allocation for the MBS sessions during the congested traffic condition only to accommodate more calls in the system. Instead of allocating fixed bandwidths for the MBS sessions and the non-MBS traffic, our scheme allocates variable bandwidths for them. However, the minimum quality of the videos is guaranteed by allocating minimum bandwidth for them. Using the mathematical and numerical analyses, we show that the proposed scheme maximizes the bandwidth utilization and significantly reduces the overall forced call termination probability as well as the handover call dropping probability.     

Evaluating the perceived voice quality on VoIP network using interpolated FIR filter algorithm

Voice over Internet Protocol (VoIP) is a popular communication service nowadays. VoIP reduces the cost of call transmission by passing voice and video packets through the available bandwidth for data packets through Internet protocol. The quality of the VoIP signal is degraded due to the various network impairments. The proposed scheme, interpolated finite impulse response, is implemented as post-processor after decoding the signal in VoIP system. The performance of the proposed scheme is evaluated for various network conditions. The results of the proposed scheme are measured with the objective measurement methods for signal quality evaluation. The performance of the proposed system is compared with the existing techniques for quality improvement in VoIP system. The results show much improvement in speech quality with the proposed scheme in comparison to other similar schemes.     

VoIP reliability: a service provider's perspective

Voice over IP services offer important revenue-generating opportunities, as well as many technical challenges in providing high-quality services. Users have come to expect highly available telecommunications services with high-quality voice. Service providers need reliable high-performance networks to meet user expectations, and must be able to guarantee performance and reliability to their customers. In converged voice and data networks, the network infrastructure must deliver very high quality and availability for some customer needs, while also providing low-cost high-capacity bandwidth for other needs. The use of quality of service mechanisms to provide prioritization for various traffic types is a key element needed for voice and data network convergence. However, it is not sufficient if the underlying networks are unreliable. The focus of this article is to address the reliability aspects of VoIP services, including the underlying IP networks.     

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.