VoIP Networks Research Articles

Voice quality prediction models and their application in VoIP networks

The primary aim of this paper is to present new models for objective, nonintrusive, prediction of voice quality for IP networks and to illustrate their application to voice quality monitoring and playout buffer control in VoIP networks. The contributions of the paper are threefold. First, we present a new methodology for developing perceptually accurate models for nonintrusive prediction of voice quality which avoids time-consuming subjective tests. The methodology is generic and as such it has wide applicability in multimedia applications. Second, based on the new methodology, we present efficient regression models for predicting conversational voice quality nonintrusively for four modern codecs (G.729, G.723.1, AMR and iLBC). Third, we illustrate the usefulness of the models in two main applications - voice quality prediction for real Internet VoIP traces and perceived quality-driven playout buffer optimization. For voice quality prediction, the results show that the models have accuracy close to the combined ITU PESQ/E-model method using real Internet traces (correlation coefficient over 0.98). For playout buffer optimization, the proposed buffer algorithm provides an optimum voice quality when compared to five other buffer algorithms for all the traces considered.

A method of interconnection between private VoIP networks and PSTN

AbstractThe use of IP telephony is spreading rapidly. It was developed chiefly for general home use, and mainly supports calls to the PSTN. In such a service configuration, various problems of realizing telephone service, such as the network address translation (NAT) problem, have generally been solved by the use of a terminal specified by the service provider. On the other hand, the design of origination/destination services between VoIP in enterprise private networks and the PSTN involves various problems, including the solution of the NAT problem, and the requirements differ considerably from those of IP telephony for general home use. This paper presents a method of solution of the technical problems related to interconnection between VoIP and PSTN in enterprise IP networks. The solution is the result of investigations emphasizing the practical aspect and repeated verification experiments, and is applicable to real commercial services. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 89(4): 42–51, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20197

Tool assisted H.323 dial‐plan design following formally defined strategies

AbstractThis paper considers H.323 v4 VoIP networks consisting of gateways and gatekeepers, addressing the task of terminating calls to multiple ISDN networks. Our goal is the maximal utilization of the gatekeeper's centralized routing mechanism for call termination to ISDN networks based on load balancing and resource availability indication (RAI), thus minimizing the need for usage of unwanted (second attempt) overflow mechanisms. We elaborate on the drawbacks of overflow mechanisms and give an overview on different solutions offered in practice by vendors. The discussion justifies the emphasis on load balancing, thus we present an algorithm and tool assisted approach for the systematic assignment of PRIs from ISDN networks to gateways. We follow strictly defined design criteria that lead to an optimized network dial‐plan. We thus offer a stand alone, off‐line solution without the need of any extra add‐ons or upgrades. This is intended to replace the purely intuitive approach followed by today's planners.Our scope is the design rather than the runtime environment; the latter continues to feature the simplicity of a single global RAI per gateway. We consider desirable and non‐desirable link distribution topologies between gateways and ISDN networks based on operational and economic evaluation criteria. These measure the number of ‘wasted’ gateway ports or ‘inappropriately’ allocated network interfaces. Our approach is illustrated through a worked out example showing key features of the proposed algorithm and revealing characteristic cases met in the ISDN interface allocation.Finally, we show how the presented methodology, which presently addresses a H.323 design issue, can provide performance benefits in the VoIP technologies of the immediate future. Copyright © 2005 John Wiley & Sons, Ltd.

Challenges in Securing Voice over IP

Although VoIP offers lower cost and greater flexibility, it can also introduce significant risks and vulnerabilities. This article explains the challenges of VoIP security and outlines steps for helping to secure an organization's VoIP network.

An Analytical Approach to Obtain the Interdeparture Time Characteristics in a Multistage VoIP Network

In this paper, we investigate the evolution of the interarrival and interdeparture times between voice packets when they are proceeding through a number of network nodes. We model the arrival process in a node as the superposition of a single tagged stream and an independent background process that aggregates the remaining traffic sources. Because we assume that the load of a single voice stream is very low compared to the load of the aggregate traffic, we can represent the tagged voice packets as markers (packets with size zero). We will establish an expression for the probability generating function (pgf) of the interdeparture time of the voice packets after one stage and use this interdeparture-time pgf as the pgf of the interarrival time of the voice packets in the next stage, in order to assess the evolution of the interarrival-time characteristics throughout the network.

Implementing VoIP: a voice transmission performance progress report

Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VoIP. This article summarizes much of what we have learned in this work, and we hope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VoIP network performance.

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.

The clearest voice [IP telephony

IP telephony is on its way to succeeding the World Wide Web as the next breakthrough application for all-IP networks. Two standards are currently competing for dominance in IP telephony - the H.323 protocol suite by the International Telecommunication Union (ITU) and the session initiation protocol (SIP) by the Internet Engineering Task Force (IETF). Both protocols exchange multimedia data over the real time protocol (RTP), so the choice of protocol suite does not influence Internet telephony QoS. However, H.323 and SIP do differ in a range of implementation aspects. The article examines these differences. It finds that SIP has clear advantages over H.323. It is more powerful, modular and extendable. It has been gaining in popularity, especially in North America and with new entrants into the VoIP market. However, H.323 has the advantage of being implemented first, which resulted in many of the early VoIP adopters embracing it. This situation makes it hard to simply conclude on one winning technology over the other. Instead, there will be a need for interoperability between SIP and H.323 signaling, for interconnection between different existing VoIP networks and PSTN/ISDN legacy deployments.

Speaking of Wi-Fi

Big organisations are realising that their wireless networks can be used for more than just data traffic. Businesses have long recognised the benefits of being able to keep in touch with employees on the move. On the face of it, the arrival of Wi-Fi provides a straightforward cost-effective solution capable of building on VoIP and wireless local area network (WLAN) technology to provide workers with the sort of communications mobility that they take for granted just about everywhere else in their daily lives. Networks that allow staff to roam with their office phone across a whole organisation are an enticing prospect, but quality of service and security are still important issues.

VoIPシステムの品質安定化制御に関する検討(マルチメディア通信とQoS, VoIP, ストリーミング, 一般)

VoIPシステムの品質安定化制御に関する検討(マルチメディア通信とQoS, VoIP, ストリーミング, 一般)

VoIPネットワークにおけるエンドポイントアドミッションコントロールの性能評価(マルチメディア通信とQoS, VoIP, ストリーミング, 一般)

VoIPネットワークにおけるエンドポイントアドミッションコントロールの性能評価(マルチメディア通信とQoS, VoIP, ストリーミング, 一般)

Endpoint Admission Control for VoIP Networks with Dynamic Routing

Endpoint Admission Control for VoIP Networks with Dynamic Routing

GSM‐IP: A VoIP service for mobile networks

AbstractBased on media gateway control protocol (MGCP), this paper describes GSM‐IP, a voice over IP service for GSM mobile network. With the signaling protocol translation mechanism provided by the MGCP signaling gateway, we demonstrate how to interwork the MGCP elements with the home location register, visitor location register and mobile switching center in the GSM network. Then we present the message flows for GSM‐IP registration, call origination, call delivery, call release and inter‐system handoff procedures. We show the feasibility of integrating GSM with the MGCP‐based VoIP network without modifying the GSM network. Furthermore, our approach does not introduce new protocol primitives to MGCP. Copyright © 2001 John Wiley & Sons, Ltd.

Admission Control for VoIP Networks using Edge-to-edge QoS measurements

Admission Control for VoIP Networks using Edge-to-edge QoS measurements

RIDE Replacement — A Hybrid PSTN/VoIP Application Solution

This paper, co-authored by BT and Telspec, describes the architecture, development and deployment of the recorded information distribution equipment (RIDE) replacement platform. The platform delivers large volumes of announcements into the PSTN yet has an internal architecture built on a multicast QoS-enabled VoIP network and provides a ready-made evolution path to terminate calls originated from IP networks. It is therefore seen as one of the first platforms to be deployed by BT to deliver consistent functionality into both the legacy PSTN and the evolving IP world.

SIP and H.323 — Interworking VoIP Networks

H.323 is currently the industry's predominant standard for offering multimedia services over IP networks, although the session initiation protocol (SIP) continues to generate significant interest in the market-place for rendezvous-based real-time applications. With its greater maturity, H.323-based equipment continues to be deployed in operational networks. However, with the advent of SIP-based third generation mobile networks and the continued development of voice over cable TV networks based upon SIP, significant deployments of SIP-based networks are on the horizon. Interworking between SIP and H.323-based equipment is of significant importance to BT and its Joint Venture partners. This paper highlights some of the issues surrounding protocol interworking between the H.323 and SIP protocol sets.It can be shown that, while possible, interworking inevitably results in sub-optimal operation and a loss of features. Thus, as interworking is inevitably going to be required for a long time to come, and the result is less than perfect, the standards bodies should be encouraged to develop service-enabling extensions to their protocols that are compatible between both protocol suites wherever possible.

Evaluating VoIP Technology — Technology Push/Market Pull

With many raised expectations of what it will offer, voice over IP has enjoyed a high profile in the communications industry. However, in common with any emerging technology, the reality of the functionality that available equipment can deliver is often at variance with such expectations. This paper highlights the role of technology assessment in refining VoIP products and highlights the value of such activities from a number of perspectives. It describes an approach to evaluate VoIP solutions, the problems associated with the testing of VoIP networks and network components, and the tools and techniques employed to do so. The paper also outlines the capabilities of the BTexaCT VoIP technology evaluation test bed, and example case studies of detailed evaluation and testing.

VoIP Applied — Lab Bench to Multimedia Service

Voice is an integral element of many of the emerging conversational IP applications found on the Internet. This paper examines the opportunities to provide these types of application at scale and considers some of the difficulties in translating what may be interesting toys on the laboratory bench into genuinely useful public communication services. It characterises early VoIP network service solutions, charting their evolution towards contemporary services and considers some of the types of application that may be possible in the future.