An Enhanced VoIP Codec Transcoder to Enhance VoIP Quality for IP Telephone Infrastructure

Poor VoIP quality in IP telephone infrastructure is a major concern and it can affect business growth, especially to businesses that deal with interacting with a client over the phone. Speech or audio signals are usually affected by codec mismatch, packet loss, and jitter which affect user perception of voice quality. VoIP telephone system is growing at a rapid speed and has received much attention because of their call cost internationally and national and fewer resources needed compared to traditional voice telephone systems or public switched telephone networks. The main aim of this paper is to develop a solution that will provide an enhanced voice quality in VoIP platform systems by implementing the amended VoIP codec transcoding system that auto negotiates VoIP codec with the intention of preventing VoIP codec mismatch via standalone and software VoIP codec transcoding system. An experimental research with technological tools such as SIP (Session Initiation Protocol) phone, asterisk PBX (Private Branch Exchange) systems and SBC (Session Border Control) will be conducted. A practical test will be carried out in any working environment with the converged network in order to test results or findings to solve the problem of codec mismatch with the intention of enhancing Voice quality and avoiding calls dropping issues in IP telephone infrastructure. This paper is introducing an amended VoIP codec transcoding system that auto-negotiate VoIP codec in order to prevent codec mismatch and enhance voice quality hence codec mismatch is not only the major concern for VoIP quality, VoIP quality can be affected by many factors, such as packet loss, jitter, packet delay, and bandwidth but this paper is focusing on the codec mismatch.