Improving the Quality of Service of Voice over Internet Protocol in Ethio Telecom Service Level Agreement Customers

Abstract

Voice over Internet Protocol is the recent communication channel and innovative service through the internet which has devoted to replacing IP network to incorporate additional value-added service like multimedia applications. VoIP permits substantial profits for both telecommunication service providers and end-users like cost savings, phone or product movability, flexibility, combined with other software or applications. However, the implementation of VoIP faces different problems like interoperability, security, and Quality of Service issues. This thesis focused on the improvement of VoIP Quality of Service problems, which are the most critical point because real-time traffic is highly sensitive to delay, packet loss, jitter, and bandwidth requirement. QoS is based on different service levels agreement in between customer and ISP network (backbone, the access, and the IP core network). Ethio Telecom signed an SLA agreement to verify guaranteed VoIP QoS with the customer but Ethio Telecom IP Network fails to fulfill the required traffic prioritization, classification, and VoIP QoS performance requirements like delay, packet loss, jitter, and bandwidth. As a result of this research gap, this thesis carried out a thorough analysis and improve VoIP QoS using BGP MPLS VPN TE and DiffServ model. Firstly, it presents a brief overview of VoIP technology. Then, it discusses the QoS issues related to real-time packet communication. Finally, develop an artifact that guarantees the real-time voice packet and QoS performance like voice packet delay, jitter, packet loss, and utilization of bandwidth. The designed artifact improves VoIP QoS performance parameters by applying BGP MPLS VPN TE and DiffServ model. DiffServ model implements a different class of service at the border of service provider Edge Router by setting traffic policing, shaping (class-based marking and policing), traffic prioritization (class-based weighted fair queue,) and congestion control technique (weighted random early discard) to improve VoIP QoS. The researcher had used a Design science Research methodology to identify data of VoIP Quality of service the problem. To Design the proposed prototype, simulation, and analysis of end-to-end VoIP QoS Architecture GNS3 and Wireshark are used, respectively. The simulation result and evaluation of the proposed end-to-end VoIP QoS Architecture show decreased packet loss, delay, jitter, and increased bandwidth utilization. which eventually boost the need of VoIP QoS Threshold parameters for SLA customer and the ITU requirement.