Performance evaluation of voice over IP using multiple audio codec schemes

Abstract

The evolution of Voice over IP (VoIP) has made it one of the most popular applications over the wired/wireless Internet system due to its flexibility in technology integration and low cost of services. Telco and service operators have used the communication resources to optimize the VoIP architecture in order to provide better quality of service (QoS) to end consumers. The VoIP is a delay-sensitive traffic which requires minimum delay for general applications and minimum loss ratio for specific applications as the key QoS performance parameters. This paper compares the end-to-end (e2e) QoS performance parameters of VoIP codec schemes against multiple traffic connections transmitted over the Internet system. Background traffics are included in the simulations to closely match the real-world Internet scenario. Simulations analysis of bidirectional VoIP communications are done from the network layer perspective to compare the QoS performances of G.711, G.729A, G.723.1 and GSM.AMR codec schemes against the incremental of active connections in the network system. The results show that the G.729A produces at least 2.81% better in term of average accumulative e2e delay. The G.711 produces at least 21.89% better in term of average accumulative e2e jitter but produces the worst e2e packet loss ratio. In addition, GSM.AMR shows the best e2e effective transmission rate ratio ranges between 42.67% and 89.82%. This study has investigated the QoS performance variations of VoIP codecs so that the results could be used as guidelines to estimate the optimal network resources for various traffic requirements as early as in the design stage. As for future works, this study suggests the adaptive priority queue and packet scheduling at Internet getaway to regulate the traffic based on per flow QoS requirements.