Improvement Of VoIP Speech Quality Based On WebRTC Platform

In this thesis,we improve the quality of voice over internet protocols(VoIP)for real-time communication.VoIP is an evolving technology that is an alternative of communication to the traditional public switched telephone networks(PSTN).However,VoIP communication is vulnerable to poor performance due to some networks impairments such as packet loss,delay and jitter.Challenges arise when trying to optimize the above impairments,in an effort to enhance the quality of speech efficiently during real-time communication.Several techniques have been used to improve quality of service(QoS)for VoIP.Congestion is mitigated by implementing congestion control methods onto the network.Network assisted dynamic adaption(NADA)addresses the congestion problem using a sender-sidebased controller.Google congestion control(GCC)is the most recently implemented congestion control method,that uses an adaptive control approach based at both the sender and receiver sides.Nevertheless,these congestion control methods can still be improved to give a better performance.In this work,we focus on developing new congestion control algorithms,that will minimize the delays,packet losses and jitter.To reduce on the packet losses,we examine different queuing management methods that are applied in real-time communication.We propose two congestion control methods.The first is based on a comparison between the initial and sending and the actual receiving rate and the second congestion control method is adaptive congestion control method,that is based on improving the sender-side of classic GCC.Our suggested method relies on extracting more information from receiver feedback reports.Therefore,our proposed algorithm is developed as an integration of efficient queuing methods into a proposed adaptive congestion control technique based at the sender-side.We conduct simulation experiments based on the network simulator(NS2)and PESQ,to test the performance of our proposed algorithms in comparison with the google congestion control method to evaluate VoIP speech quality.Furthermore,when this proposed algorithm is compared to the existing methods,results indicate that it outperforms them.