Modules for implementing a voice over internet protocol telephony system on a TMS320C6711 DSK using an LD-CELP voice CODEC algorithm

• Main Authors: Cabanlit, Mark Andreu H., Lipa, Alfredo G., Olegario, Rolan P., Panugayan, Joel L., Salvador, Vanessa Lourdes T.
• Format: text
• Language: English
• Published: Animo Repository 2002
• Subjects: Engineering
• Online Access: https://animorepository.dlsu.edu.ph/etd_honors/169
• Institution: De La Salle University

Abstract. The emergence of the trend in globalization called for a need to establish a low cost communication system that would allow the transfer of both data and voice. Current technology offers that there is a separate network for data and voice, which is costly due to the fact that maintenance is expensive. Already, VoIP is gaining ground in the industry as an alternative communication tool as this allows the users to maintain only a single network for both voice and data. The basic idea of VoIP is that it compresses voice and allows it to travel in the data network, this is in turn allows users of voice and data to share a common medium, maximizing the efficiency of the network. Currently, there are several proponents working on Client software solutions for VoIP. However, software based VoIP would not be able to provide enough advantage as compared to embedded system VoIP because the embedded system frees a computer workstation to perform other task when in conference. This in turn would mean better savings and efficiency for corporations. This study investigates the application of VoIP into TMS320C6711 DSK, with the use of an LD-CELP voice codec algorithm that compresses a 64 kbps speech sample into a 16 kbps codevector representation would allow transfer of same amount of information is a smaller bandwidth.