Performance Analysis of various Codecs using RSVP on VoIP Quality of Service over Variable Bandwidth

With the rapid growth of the Internet, customers are demanding multimedia applications such as telephony and video on demand, to be available on the internet. Voice over Internet Protocol (VoIP) is a rapidly emerging technology for voice communication that uses the ubiquity of IP-based networks. The greatest challenge is the provisioning of Quality of Service (QoS) over limited bandwidth. Multimedia applications need a lot more bandwidth and have different QoS requirements than the applications that were used in early years of the internet. The purpose of this paper is to check the performance of VoIP application under different Codec’s such as G.711, G.729 and G.723.1 over variable bandwidth. This paper proposes RSVP protocol for providing end to end QoS for real time applications over diverse networks. The performance is checked under RSVP and Non RSVP. Packet end to end delay and packet delay variation are evaluated through simulation. The result shows that under any bandwidth G.723.1 gives better result as compared to G.711 and G.729. Simulation is done using OPNET IT Guru Academic edition.