SIP-Based QoS in IP Telephony

Abstract

In this paper, the authors analyze the factors resulting to the degradation of the quality of service in voice over IP (VoIP) telephony. SIP protocol is used to explore the QoS characteristics of different Codec. The system is designed for an IP telephony service provider with other GSM operator network cloud. A soft IP PBX maintains the dial pattern, SIP proxies are used in order to implement call control and to allow the distributing of the gateways’ lines. The system is executed in a test bed where QoS factors likes delay, packet loss, forward and reverse jitter, MOS and delta measured. A Popular simulation software Wireshark is used to simulate and analysis the transmission characteristics of packetized voice over the IP network channel. The results are graphically shown, which reveal the effects of packet size on QoS. Packet sizes from 10 to 60 bytes are used for a G.729 Codec in 11.2-24 Kbps channel bandwidths are studied. A 20 byte packet size at 24 Kbps channel bandwidth is seen to show the best result in terms of jitter, delay and delta. The results of studies conducted by also shown that G.729 codec shown high MOS value and low mean forward and reverse jitter for simultaneous call compare with other codecs