A new adaptive redundancy control algorithm for VoIP applications (original) (raw)

2013, 2013 IEEE Global Communications Conference (GLOBECOM)

Dynamic Adaptation of Quality of Service for VoIP Communications

2009

The present work proposes an adaptive solution to provide quality of service in Voice over IP communications. This solution is based on three components that interact in order to achieve higher quality in voice communication. The first two consist in changing the codec and the transport protocol in real-time during a conversation; the third consists in using a Forward Error Correction mechanism to recover from loss packets. To demonstrate the voice quality obtained by this solution, a VoIP client application was developed, compatible with other VoIP clients, to implement the proposed quality of service algorithm and control the voice quality during a conversation. The results of the experimental measurements and simulations performed demonstrate that this solution is viable and significantly increases the voice quality of VoIP communication.

Employing Mean Opinion Score of Audio Lossy Compression Algorithms in VoIP Application

Communication is vital. It is constructed from one individual to a group of people or vice versa. Nowadays, people always want to be connected anytime and anywhere while using communication devices. Thus, there are great demands of VoIP applications that offer quality audio and low resource utilization. In this study, some audio compression algorithms also known as codecs are considered and integrated into VoIP application. Computation of jitter, packet loss, and bandwidth is examined and analyzed. Also, the G.107 Mean Opinion Score is applied to measure the scalar quality rating value of voice using different codecs in VOIP Application.

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