Performance Evaluation of G.711 and GSM Codecs on VoIP Applications Using OSPF and RIP Routing Protocols

Abstract

In recent years, Voice over Internet Protocol (VoIP) has quickly become one of the most powerful and popular technologies for communication. From business phones to social messaging applications with millions of users, VoIP is an underlying technology that powers the way each other connects across the world. For this reason, establishing an excellent audio quality has become a challenge for VoIP applications. There are several factors in determining the effective utilization and quality of VoIP solutions, namely: echo control, packet loss, delay variation, jitter, network topology, and more importantly, the choice of voice encoding codec. Choosing the right VoIP codec can make an impeccable difference in voice quality. In this chapter, the behavior of two different voice encoding codecs, namely G.711 and GSM, is evaluated using the Riverbed Modeler 17.5. Simulation results show that G.711 yields a better performance compared to GSM in terms of traffic sent, traffic received, jitter, end-to-end delay, packet delay variation, and mean opinion score (MOS), chosen as performance metrics, indicating that G.711 transmits a better audio quality than GSM in VoIP applications.