#TITLE_ALTERNATIVE#

• Main Author: WIRAWAN (NIM 13204026), FIKRI
• Format: Final Project
• Language: Indonesia
• Online Access: https://digilib.itb.ac.id/gdl/view/10448
• Institution: Institut Teknologi Bandung

Internet telephony is an upcoming and very promising technology that enables people to make telephone calls via an IP network such as the internet. One of the important issues is Quality of Services (QoS) of internet telephony. A large number of factors are involved in making a high-quality of services. These factors include the delay, jitter, packet loss, and speech codec.<p> <br /> <br /> <br /> <br /> <br /> The objective of this research to understand the quality of internet telephony which is acceptable for most users based on network parameters such as delay and packet loss at low speed network which has 64 Kbps, 128 Kbps, and 256 Kbps of network speed. This research also to understand the best codec and jitter buffer value that applicable for internet telephony on low speed network.<p> <br /> <br /> <br /> <br /> <br /> In this research, we build a testbed network of internet telephony system on low speed network. We generate some voice call to the testbed network which has different network parameters. We obtain the voice quality by comparing voice at sender and receiver.<p> <br /> <br /> <br /> <br /> <br /> The measurements show that maximum calls achieved by applying codec G.723.1 5.3 Kbps. The quality of speech is recommended when packet loss below 7% for codec G.729; below 4.5% for codec G.723.1 5,5 Kbps; below 5.5% for codec G.723.1 6.3 Kbps, and when delay below 100 ms and jitter below 30 ms for codec G.729, G.723.1 5,3 Kbps, and G.723.1 6,3 Kbps. The appropriate jitter buffer value for internet telephony system on low speed network is 50ms.