Behavior analysis of VoIP performances in next-generation networks

In the last decade, the telecommunications industry has evolved. Traditional networks cannot meet customers' incessant demand for new multimedia and interactive services. Next-generation networks (NGN) have been proposed to improve the quality of service (QoS) of the transported traffic as well as to make the network evolving, thus supporting future applications. Multi-Protocol Label Switching (MPLS) and IPv6 are used mainly on transport layer of NGN. These provide higher reliability, QoS, and security levels for the user’s applications. Voice over IP (VoIP) is one of those applications that gaining momentum in our daily lives. Individuals or organisations tend toward this type of services given their various benefits. Several factors can degrade the quality of VoIP, such as the number of established sessions, the Call Signalling Protocol (SIP or H.323), the used transport protocols (IPv4 or IPv6) and especially the VoIP codecs chosen (GSM, G71, and G729). According to our research, several works have been carried out evaluating the performance of VoIP on traditional networks. However, in this paper, we will assess the performance of the VoIP application in NGN architecture with MPLS as the transport protocol. The evaluation takes into consideration: (i) the impact of transport protocol on call establishment duration, (ii) the effect of user number and used codec on VoIP quality. The evaluation parameters used are the duration of the establishment of the call by the SIP protocol, the MOS score, latency, loss rate, and CPU. The results obtained showed that the call signalling duration depends mainly on the transport protocol in addition to the network architecture. The number of clients influences the quality of VoIP. However, the used codec may allow a higher number of VoIP sessions to be established.