Edjair Mota

An E-model implementation for speech quality evaluation in VoIP systems

This article presents a voice quality measurement tool based on the ITU-T E-model. Firstly, the ITU-T and ETSI specifications of E-model are briefly reviewed and some errors found in these documents are pointed. After, a measurement tool based on the corrections is described. VoIP calls through the Brazilian National Education and research network (RNP) backbone were used to verify the tool operation.

Synchronizing web browsing data with Browserver

People spend a lot of time navigating on the web. When moving from one computer device to another, it would be useful to have access to the navigation data produced in the previous web session. In this article, a synchronization service of navigation data, called Browserver, is presented. It is responsible for keeping user navigation information (tabs, history, forms, cookies, etc.) so that it can be recovered from any device connected to the Internet. Finally, Browserver performance is compared to a similar service, according to hardware and network consumption metrics.

Can I add a secure VoIP call

Voice over IP is a major trend in applications for wireless networks, but even so it is not immune to the risks usually related with IP networks. Proposed solutions for VoIP security are already in the market, but these solutions must take into account the real-time constraint of voice service and their mechanisms should address possible attacks and overhead associated with it. One of these solutions is to use IETF IPSec to guarantee confidentiality in order to address security design holes of wireless VoIP networks. This article performs an experimental comparison of the impact of encryption mechanisms on voice speech quality in widely deployed wireless technologies: 802.11 and Bluetooth. Evaluates the upper bound on number of simultaneous VoIP calls which can be placed in a single cell of both networks when security is applied and uses the computational model E-Model to assess quantitatively the quality of service

GOP size influence in high resolution video streaming over wireless mesh network

Video transmission over IP networks requires appropriate conditions of quality of service, in order to ensure the integrity of the image, which still represents a challenge to be overcome in wireless mesh networks due to traffic capacity limitations inherent to its architecture. In this work we evaluate the performance of high-resolution video over multi-hop mesh networks and we propose a suitable GOP size for encoding video to be streamed, based on traffic conditions to minimize the loss of quality. Experiments were carried out in a wireless mesh network planned for real testing within a 1500m2 area of the main UFAM campus, using the IEEE 802.11a and 802.11n network standards.

A framework for network quality monitoring in the VoIP environment

Monitoring speech quality in Voice over IP (VoIP) networks is important to ensure a minimal acceptable level of speech quality for IP calls running through a managed network. Information such as packet loss, codec type, jitter, end-to-end delay and overall speech quality enables the network manager to verify and accurately tune parameters in order to adjust network problems. The present article proposes the deployment of a monitoring architecture that collects, stores and displays speech quality information about concluded voice calls. This architecture is based on our proposed MIB (Management Information Base) VOIPQOS, deployed for speech quality monitoring purposes. Currently, the architecture is totally implemented, but under adjustment and validation tests. In the future, the VOIPQOS MIB can be expanded to automatically analyze collected data and control VoIP clients and network parameters for tuning the overall speech quality of ongoing calls.

A model for contact volume prediction in DTNs

Delay and Disruption Tolerant networks (DTNs) are an alternative means of providing Internet access to remote communities, where there is a lack of a telecommunications infrastructure. Communication in this scenario is carried out through opportunistic encounters between nodes, called contacts, in which messages are forwarded to the destination. Predicting information about a contact, at the time of its occurrence, allows strategies to be adopted for optimizing message dispatch to improve the delivery rate and reduce the delivery delay in DTNs. This paper outlines a model to predict the amount of data that can be transferred, during a DTN encounter, based on real experiments conducted in the Amazon region. The results show that the proposed model has the advantage of including environmental conditions, such as soil and vegetation in the Contact Volume prediction, plus parameters like Radio Frequency power, the wireless networking standard, and the type and height of the radio antennas.

Survey on application-layer mechanisms for speech quality adaptation in VoIP

VoIP calls are sensitive to several impairments, such as delay and packet loss. One way to overcome these problems is by adaptively adjusting application-layer parameters to keep a minimum speech quality level. At the heart of self-adaptive systems lies a feedback loop, which consists of four key activities: monitoring, analysis, planning, and execution. Nevertheless, the existing adaptive approaches to QoS control of VoIP do not explicitly exhibit this feedback loop. Bringing it to surface can help developers in designing more robust and human-independent VoIP systems. This survey presents a comprehensive review of the current state-of-the-art research on speech quality adaptation of VoIP systems at the application layer and some research challenges on this subject.

Towards an efficient header compression scheme to improve VoIP over wireless mesh networks

Header compression techniques can improve the speech quality of VoIP applications. However, its deployment over wireless mesh environments may not offer much improvement, since the wireless and multi-hop characteristics of such a network impose a high packet loss rate. In this work, we propose a new approach for header compression, using packet aggregation to cooperate with the header compression scheme. Simulation results show that the new cooperative approach can decrease significantly the packet loss, impacting positively on the speech quality.

Analysis of the secure RTP protocol on voice over wireless networks using extended MedQoS

This paper presents an empirical investigation of the impact of Secure RTP (SRTP) on VoIP calls over wireless networks: 802.11 and Bluetooth. For the purpose of evaluating this impact we developed an analysis tool based on E-Model and security aspects of SRTP which attempts to determine the balance of quality of service versus security. The results demonstrate that the impact of SRTP to VoIP should not be disregarded. MOS scores of secure calls computed from E-Model pointed out an undesirable level of quality of service on single calls using wireless channels. This quality degradation also leads to the reduction of channel capacity to offer simultaneous calls. The developed analysis tool indicated three important factors to this degradation: time to encrypt, time to authenticate and time to form a cryptographic context for each SRTP packet sent and received.

Header compression for VoIP over multi-hop wireless mesh networks

Header compression is both an interesting and a challenging alternative to real time applications. This paper presents an evaluation of a header compression technique called CoHC - cooperative header compression - and its performance in Voice over IP within multi-hop wireless mesh networks. The CoHC technique is able to perform header compression over multiple channels in a cooperative manner in order to avoid loss of synchronism between the compressor and the decompressor. Simulation results showed that the evaluated scheme can impact positively on VoIP calls quality.