To improve the voice quality over IP using channel coding

Abstract

Voice over Internet protocol (VoIP) is the transmission of voice over networks using the internet protocol. IP networks have become increasingly popular in the past few years, due to the exponential growth of the public internet leading the way in to the IP world. Long distance calls, especially international subscriber dialing (ISD), can be made significantly less expensive when supported by an IP network rather than by the PSTN. Any call made is supported by VoIP technology and it involves the transmission of many individual packets over an IP network. Thus the cost of VoIP calls in part depends on the number and size of packets that must be transmitted. So voice (source) compression technology is used to reduce the amount of bandwidth required in order to reduce cost and to reduce the delay impact from network. But, compression techniques increase the network impairments also. When packets are transmitted through network they are affected by impairments, like packet drop and end-to-end delay. The main agenda of VoIP service providers is to provide good quality of service (QoS). The objective of this research work done is to minimize, the error introduced in channel due to the above mentioned network impairments and hence improve the quality of sound, and also analyze the voice quality with and without use of channel coding scheme. There are two measuring techniques, subjective and objective. In subjective method we measure the mean opinion score (MOS) and use matching algorithms in objective methods to measure the difference between decoded outputs of source coding system and joint source-channel coding system. In this work, the objective method is used and it is observed that the performance of the system improves significantly