PENGEMBANGAN SISTEM VOIP BERBASIS ASTERISK DENGAN CO-DEC OPUS UNTUK SIARAN LAPANGAN

    Ahmad Ali Yasin, - and Iwan Kustiawan, - (2025) PENGEMBANGAN SISTEM VOIP BERBASIS ASTERISK DENGAN CO-DEC OPUS UNTUK SIARAN LAPANGAN. S1 thesis, Universitas Pendidikan Indonesia.

    Abstract

    Penelitian ini mengatasi keterbatasan sistem komunikasi tradisional seperti Studio Transmitter Link (STL) dan Public Switched Telephone Network (PSTN) dalam siaran lapangan, yang mendorong adopsi Voice over Internet Protocol (VoIP) berbasis Asterisk. Tujuan utama penelitian adalah mengevaluasi performa codec Opus dibandingkan codec tradisional (G.711 A-law, G.711 U-law, G.722, GSM-FR) dalam hal kualitas audio, serta menganalisis kemampuan server Asterisk menangani komunikasi suara waktu nyata dalam berbagai kondisi jaringan. Metode penelitian melibatkan implementasi sistem VoIP berbasis Asterisk dengan codec Opus, diikuti pengujian Quality of Service (QoS) meliputi jitter, packet loss, bandwidth, dan latency. Kualitas suara dievaluasi secara objektif menggunakan PESQ (ITU-T P.862) dan subjektif melalui MOS (ITU-T P.800) dengan 25 partisipan, dalam tiga skenario jaringan: stabil, padat, dan sangat padat. Hasil menunjukkan codec Opus unggul dalam kualitas suara, mencapai skor MOS-PESQ rata-rata 3.125 (kategori 'Baik') dan MOS subjektif 4.28 (kategori 'Sangat Baik'). Opus juga menunjukkan ketahanan terhadap packet loss (rata-rata 0.184%) meskipun dengan konsumsi bandwidth tertinggi (rata-rata 144.62 kbps). G.722 menempati posisi kedua dengan MOS-PESQ 2.929 dan MOS subjektif 3.76. GSM FR menunjukkan kualitas terendah (MOS-PESQ 1.962, MOS subjektif 3.20) namun paling efisien bandwidth (rata-rata 28.88 kbps). Analisis korelasi Pearson antara PESQ dan MOS menunjukkan hubungan sangat kuat (r = 0.945, p < 0.05), memvalidasi PESQ sebagai prediktor andal. Dalam skenario jaringan padat dan sangat padat, performa semua codec menurun, namun Opus dan G.722 menunjukkan ketahanan lebih baik. Penelitian ini menegaskan superioritas Opus untuk aplikasi VoIP yang membutuhkan kualitas suara tinggi dan adaptabilitas jaringan. This research highlights the limitations of traditional communication systems like the Studio Transmitter Link (STL) and the Public Switched Telephone Network (PSTN) in field broadcasting, advocating for Asterisk-based Voice over Internet Protocol (VoIP) systems instead. The primary aim is to evaluate the performance of the Opus codec compared to traditional codecs (G.711 A-law, G.711 U-law, G.722, and GSM-FR) in terms of audio quality while assessing the Asterisk server's ability to manage real-time voice communication across various network conditions. The methodology involves implementing a VoIP system based on Asterisk using the Opus codec, followed by Quality of Service (QoS) testing, which measures jitter, packet loss, bandwidth, and latency. Voice quality is evaluated both objectively, using the Perceived Evaluation of Speech Quality (PESQ, ITU-T P.862), and subjectively through the Mean Opinion Score (MOS, ITU-T P.800) with 25 participants in three network scenarios: stable, congested, and highly congested. Results show that the Opus codec excels, achieving an average MOS-PESQ score of 3.125 (Good) and a subjective MOS of 4.28 (Very Good). Opus exhibits low packet loss (0.184%) but high bandwidth consumption (144.62 kbps). G.722 ranks second in quality, while GSM-FR proves bandwidth-efficient but scores the lowest in audio quality. Correlation analysis reveals a strong relationship (r = 0.945, p < 0.05), confirming PESQ as a reliable predictor. Overall, this research confirms Opus's superiority for VoIP applications requiring high audio quality and network adaptability.

    Baca Full Text klik disini
    [thumbnail of S_TE_2107301_chapter1.pdf] Text
    S_TE_2107301_chapter1.pdf
    Download (1MB)
    [thumbnail of S_TE_2107301_chapter2.pdf] Text
    S_TE_2107301_chapter2.pdf
    Restricted to Staf Perpustakaan
    Download (8MB)
    [thumbnail of S_TE_2107301_chapter3.pdf] Text
    S_TE_2107301_chapter3.pdf
    Download (5MB)
    [thumbnail of S_TE_2107301_chapter4.pdf] Text
    S_TE_2107301_chapter4.pdf
    Restricted to Staf Perpustakaan
    Download (7MB)
    [thumbnail of S_TE_2107301_chapter5.pdf] Text
    S_TE_2107301_chapter5.pdf
    Download (1MB)
    [thumbnail of S_TE_2107301_Appendix.pdf] Text
    S_TE_2107301_Appendix.pdf
    Restricted to Staf Perpustakaan
    Download (6MB)
    [thumbnail of S_TE_2107301_Title.pdf] Text
    S_TE_2107301_Title.pdf
    Download (5MB)
    Official URL: https://repository.upi.edu/
    Item Type: Thesis (S1)
    Additional Information: https://scholar.google.com/citations?hl=en&user=tiS5L30AAAAJ ID SINTA Dosen Pembimbing: Iwan Kustiawan: 5996452
    Uncontrolled Keywords: VoIP, Asterisk, Opus, PESQ, MOS, Kualitas Suara, Jaringan. VoIP, Asterisk, Opus, PESQ, MOS, Sound Quality, Network.
    Subjects: T Technology > T Technology (General)
    T Technology > TA Engineering (General). Civil engineering (General)
    T Technology > TK Electrical engineering. Electronics Nuclear engineering
    Divisions: Fakultas Pendidikan Teknik dan Industri > Jurusan Pendidikan Teknik Elektro
    Depositing User: Ahmad Ali Yasin
    Date Deposited: 05 Nov 2025 06:43
    Last Modified: 05 Nov 2025 06:43
    URI: http://repository.upi.edu/id/eprint/144971

    Actions (login required)

    View Item View Item