Mirko Luca Lobina

Speech playout buffering based on a simplified version of the ITU-T E-model

In Internet-protocol (IP) telephony, problems of transmission delay variations are frequently addressed with adaptive dejitter buffering techniques. These are aimed at setting the buffer dimension so as to limit the packet end-to-end delay, the total packet loss, or both together. The selection of delay and loss limits is of key importance for the resulting conversational quality. This problem is addressed in this letter, whose main contribution is the introduction of a perceptually motivated optimality criterion that allows the receiver to automatically balance packet delay versus packet loss. In the proposed approach, the dejitter buffer size is adaptively set, and the adopted criterion relies on the use of a simplified version proposed by Cole and Rosenbluth of the conversational-quality International Telecommunication Union (ITU) Telecommunication Standardization Sector (ITU-T) E-Model.

Playout buffering in ip telephony: a survey discussing problems and approaches

Playout buffering is a key component in telephony over IP services. It allows the receiver to compensate network jitter and to resynchronize the received packets flows so as to achieve smooth voice decoding. The management of the playout buffer heavily influences the final quality of service (QoS), since the packet loss rate and total one-way delay depend upon it. In this article, starting from an analysis of the jitter impairment in real-time voice communications, we survey the approaches that have been proposed in the past to address this problem. A classification and a chronological timeline of strategies and breakthroughs are provided, with an analysis of interactions with other related areas. Particular attention is devoted to the techniques employing sophisticated speech quality models, which represent the latest advance in this field. The article ends with an analysis of the advantages and weaknesses of the proposed solutions. PLAYOUT BUFFERING IN IP TELEPHONY: A SURVEY DISCUSSING PROBLEMS AND APPROACHES 3RD QUARTER 2006, VOLUME 8, NO. 3 www.comsoc.org/pubs/surveys

Playout buffering of speech packets based on a quality maximization approach

To combat jitter problems in voice streaming over packet networks, playout buffering algorithms are used at the receiver side. Most of the proposed solutions rely on two main operations: prediction of delay statistics for future packets; setting of the end-to-end delay so as to limit or avoid packet losses. In recent years, a new approach has been presented, which is based on using a quality model to evaluate the impact of both packet loss and delay on the voice quality. Such a model is used to find the buffer setting that maximizes the expected quality. In this paper, we present a playout buffering algorithm whose main contribution is the extension of the new quality-based approach to the case of voice communications affected by bursty packet losses. This work is motivated by two main considerations: most of IP telephony applications are characterized by bursty losses instead of random ones; the human perception of the speech quality is significantly affected by the temporal correlation of losses. To this purpose, we make use of the extensions proposed in the ETSI Tiphon for the ITU-T E-Model so as to incorporate the effects of loss burstiness on the perceived quality. The resulting playout algorithm estimates the characteristics of the loss process varying the end-to-end delay, weights the loss and the delay effects on the perceived quality, and maximizes the overall quality to find the optimal setting for the playout buffer. The experimental results prove the effectiveness of the proposed technique.

Playout buffering in IP telephony: a quality maximization approach

In IP telephony, playout buffering is operated at the receiver side to compensate for network delay variations. This function allows the receiver to reduce packet lateness at the expense of an increase in the end-to-end delay. In recent years, a new approach has been proposed for playout buffering. It is based on using a voice quality model capable of predicting the end-user perceived quality from the network and system parameters. The playout delay (end-to-end delay) is set so as to maximize the quality according to the adopted model. In this paper, we propose a new algorithm based on this approach. The major contribution is the ability to deal with the burstiness in packet losses. This is a significant advance for two main reasons: bursty losses often affect IP telephony applications; the temporal loss correlation influences the human perception of the quality degradation. Accordingly, we propose a new optimality criterion to control the playout delay, which is based on the extensions proposed by the ETSI Tiphon for the ITU-T E-model. Experimental results on real traces prove the efficacy of the proposed algorithm.

A Secure MPLS VPN Infrastructure for Complex Geodata Sensor Network

A complex geodata sensor network is the sum of several sensor networks, each of which composed of different typologies of sensors monitoring specific environmental factors. The main application of complex geodata sensor network is collection and forecast of geo-information. To achieve accurate forecasts at high spatial accuracy, several single networks of sensors are interconnected together so that homogeneous data coming from different sources are collected and processed in real-time. This work focuses on the problem of interconnecting different networks together and to a central system of data storage. Specifically, an MPLS approach is proposed for guaranteeing good results in terms of overall security and bandwidth.

IP telephony over satellite networks: control of the playout delay to maximize the perceived quality

In streaming applications, playout buffering is operated at the receiver side to compensate network delay variations. This function allows the receiver to reduce packet lateness at the expenses of an increase in the end-to-end delay. IP Telephony is one of the applications where this function is crucial since transmission delays heavily affect the interactivity and significant losses injure the signal quality. Many approaches have been proposed in the last decade to control the playout delay in IP Telephony, where the most recent ones rely on the use of the ITU-T E-Model to evaluate the quality perceived by the end-user. However, none of these specifically deals with the scenario where the service is provided over satellite networks. Indeed, nowadays, this scenario is quite frequent with the reduction of the costs of the satellite channels and equipments. In this case, there are some specific issues that need to be dealt with to improve the effectiveness of the playout control algorithms. These issues are related to the delays that are encountered when satellite links are involved. These aspects are described in this paper, which also presents a new quality-based playout algorithm that provides better results with respect to alternative approached when satellite transmissions are involved.