#TITLE_ALTERNATIVE#

• Main Author: HENRY WIJAYA (NIM 13203041); Pembimbing : Ir. Armein Z. R. Langi M.Sc., Ph.D. dan Dr. Yoan, CHRISTIAN
• Format: Final Project
• Language: Indonesia
• Online Access: https://digilib.itb.ac.id/gdl/view/15229
• Institution: Institut Teknologi Bandung

Nowadays the amount of Internet user is growing very fast. It is about approximately 1.4 billion or 21.9 % of the world population. One of the Internet applications that is growing very fast is Voice over Internet Protocol or VoIP.<p> <br /> <br /> <br /> VoIP is an application that allows the user to do a two-way conversation using Internet network. The main benefit of VoIP is its low price. VoIP also has benefits that Internet has, such as data tranfer, and the other applications. In other word, VoIP is a multi application.<p> <br /> <br /> <br /> Besides benefits, VoIP also has drawbacks. The main issue of VoIP application is the sound quality of Quality of Service (Qos) it could provide. In voice transfer process, VoIP convert the voice that is in analog state into voice packets that are delivered through the Internet network. These packets maybe delayed or even lost in its way to the destination terminal. These become the main parameters in VoIP application: delay, jitter, and packet loss. The codec used and network capacity also affect the Quality of Service.<p> <br /> <br /> <br /> This final assignment report discuss briefly about the correlation between delay, jitter, codecs, packet loss and the sound quality of VoIP. Here, research is done by sending sound signal from client terminal to server terminal through routers using a program called SIPp. This program simulates several calls simultaneously in SIP protocol. The parameters is simulated using traffic controller that is provided in Linux operating system. The signal sound received in the client terminal is compared to the original sound signal using Perceptual Evaluation for Speech Quality (PESQ) method and then converted into Mean Opinion Score (MOS).<p> <br /> <br /> <br /> By altering some parameters affecting the characteristic of the examined IP network's structure, we are able to construct relational graph among its parameters and speech compression used regarding the quality of the sound. Based on these graphs, we can then analyze how great the quality reduction within each protocol and speech compression. Quantitative data resulted here can be used as references or comparators to establish speech's quality and the IP network's parameters constraints of the network where VoIP communication technologies are to be used appropriately.