In recent years, research on VoIP over wireless mesh network (WMN) has gained particular attention because of its commercial prospect. This paper presents an analytical method to estimate VoIP call capacity in an WMN employing IEEE 802.11 devices. We used Markov chain analysis of IEEE 802.11 for network delay and loss to estimate the capacity while using rating factor, R score, defined by ITU-T, to ensure call quality. A detailed analysis of queueing delay and loss in terms of network parameters is also carried out along with their impacts on voice quality. The capacity model estimates call capacity in a single hop WLAN and is extendable to multi hop scenario and for video communications. The theoretical results are verified by simulation and compared to related previous works.

VoIP Call Capacity over Wireless Mesh Networks

Voice, in the form of VoIP, has emerged as a dominating traffic category in the Internet. VoIP traffic suffers from severe voice quality degradation if the end-to-end delay and loss becomes high. PCF medium access mechanism offers delay bound delivery which is a crucial attribute for delay sensitive applications like VoIP. But due to its unavailability in most commercial products, little analysis has been done on PCF for supporting VoIP traffic. Most importantly, standard voice quality assessment methods are not used in the capacity studies. We propose a VoIP capacity model for PCF that uses voice quality as the limiting factor and takes imperfect channel condition into account. The effects of aggregation level, packet error rate and voice quality on VoIP capacity are investigated, and a comparison between voice capacities under PCF and DCF is presented. The proposed model will be extremely useful in designing networks to support quality voice calls.

VoIP Capacity over PCF with Imperfect Channel

Current networks are multi-service packet-switched environments, very often based on Internet Protocol, where the provision of good quality services is a great challenge. The most popular service, offered by almost every network, is voice transmission. As a real-time service, it places high demands on bandwidth and delay. The paper describes selected network dimensioning problems for the voice over IP service. The authors discuss service quality measures in legacy circuit switched networks and the key performance indicators that influence the quality of telephone service in packet networks. The main contribution of the paper is the proposal of utilizing the blocking probability calculation, which is based on the B-Erlang model used in legacy systems, to estimate the average bandwidth needed to handle a number of sources generating voice traffic in IP networks. Taking into account the average bandwidth required to carry out a one voice conversation, the total network capacity was calculated. The authors made analytical calculations which were confirmed by simulation studies. The problem of using different codecs and their influence on bandwidth consumption and overall service quality experienced by a user were also discussed in the paper. At the end of the paper, the authors defined the directions of further research that should be carried out, i.e. the development of dimensioning methods for networks carrying multimedia services with particular attention paid to video transmission and reliable quality assessment.

Quality-aware network dimensioning for the VoIP service

International standardization organizations, responsible for preparing specifications (such as IMT-Advanced) for emerging 4G networks, define requirements for system level simulations for the candidate technologies [1], [2]. The goal behind those documents is to facilitate System-Level-Simulations by providing common methodology to perform such simulations (i.e. for WiMAX). According to [1] cell-level simulations can be an intermediate step between Link and System-level simulation where the capacity of a single cell and a single Base Station, providing service for multiple users, is evaluated by means of comprehensive tests. Still the IEEE standardized simulation methodology [1] does not specify how to evaluate (WiMAX) system capacity with various connection admission control mechanisms. Therefore as a first step we focus on the problem of adjusting simulation methodology to facilitate simulations covering CAC with Time Division Multiplexing Access scheme (TDMA), OFDM and uplink traffic. The applied evaluation methodology is derived from the best-practices in IEEE 802.16m Evaluation Methodology Document and WiMAX Forum’s System-Level-Simulation (SLS) methodology. Afterward the introduced methodology is utilized to find answers to the following problems:  To what extent does the capacity change when different FEC codes are deployed (Convolutional Turbo Coding CTC, non binary Low-Density Parity-Check nbLDPC)  What is the user perception of the service quality (Quality of Experience QoE) and what are the differences in the system performance when different FEC codes are deployed?  How to improve resource estimation, especially when considering connection requests arriving in large batches?  How the performance of traffic – aware admission control algorithms changes, when some users follow VoIP traffic pattern with silence-suppression enabled?  Does the performance of measurement based CAC change, if the system experiences situations, in which connection requests arrive in large batches?

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Evaluation of QoS and QoE in Mobile WIMAX – Systematic Approach

The authors contribute to the spectrum of quality of experience (QoE) research by defining simulation platform for evaluation of VoIP quality of experience over a wireless WiMAX link. In particular the authors show how the Matlab/Simulink and ns2 integration can be facilitated to tackle the QoE aspects in a comprehensive manner. Furthermore the process of measuring QoE using an off-line implementation of the ITU-T E-model is presented and applied to the results of the QoS test campaign. Simulations are performed with the use of robust nb-LDPC coding scheme, which is current research topic in the FP7 DaVinci project. The obtained results of QoE over WiMAX are later compared to the results for Red-Solomon Convolutional Codes.

QoE assessment of VoIP over IEEE 802.16 networks with DaVinci codes using E-model

Quality of service for Voice over IP on wireless links (WiFi) has to be measured by the perceptual speech quality. How can this subjective measures be "captured" by models that rely on objective measures based on transmission delays and packet losses monitoring. These parameters which are common on such links can affect dramatically speech quality. In this paper, The ITU-T legacy e-Model, recently proposed to qualify audio and well adapted to circuit switched networks is adapted for packet switched networks. A new model to automatically quantify or measure the speech quality perception for voice/audio communications over IP. named here "packet-e-Model", adapts the e-Model to packet networks in general with a special emphasis on WiFi.

A. Meddahi

Papers

"Packet-e-model": e-model for wireless VoIP quality evaluation