scholarly journals QoS measurement of VoIP codec usage on limited bandwidth network over UDP-based VPN

2020 ◽  
Vol 12 (1) ◽  
pp. 13-17
Author(s):  
Panca Susanto
Keyword(s):  
Packet Loss ◽  
Sound Quality ◽  
Communication Quality ◽  
Voice Communication ◽  
Limited Bandwidth ◽  
Delay Jitter ◽  
Opinion Score ◽  
Tunneling Method ◽  
Encryption Method ◽  
The Voice

The usage of internet protocol for voice communication is widely used and more efficient rather than an analog signal. However, there is no security guaranteed on IP-based voice communication. The voice payload can be easily tapped or even manipulated. In the case of improving the security aspect, communication quality should be also considered. VoIP requires sufficient bandwidth to get proper communication quality. The ITU-T released a standard unit of communication quality, known as Mean Opinion Score (MOS) which is made from the subjective judgments of some individuals. However, MOS method takes time and is expensive. In this research, we measure VoIP communication which is secured by using VPN and build a tool for analyzing the voice packet between communication peers. The tool has capabilities to measure delay, jitter, and packet loss. Since VoIP has a QoS standard by ITU-T, the usage of VPN for security purpose needs to be considered. The sound quality might be decreased due to the addition of header for tunneling method, as well as the additional delay when the encryption processing is carried out. We used 3 types of codec: a-Law, GSM, and iLBC which will be passed on 4 types of bandwidth (256, 128, 64, 32 kbps) through the UDP-based VPN that use 3 types of encryption method (3-DES, Blowfish, AES).

scholarly journals Analisis Perbandingan Performa VPN IPSec dan ZRTP pada VoIP

2018 ◽  
Author(s):  
muhammad iqbal ◽  
Galih Yogi Noviantoro ◽  
Amarudin
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Peer To Peer ◽  
Mean Opinion Score ◽  
Total Delay ◽  
Delay Jitter ◽  
Diffie Hellman ◽  
Opinion Score

Protokol ZRTP berperan melakukan enkripsi data pada sistem komunikasi VoIP, serta adanya suatu teknologi jaringan komputer VPN IPsec dengan sistem melewatkan data dalam suatu ip private virtual atau sebagai tunnel untuk media transmisi data yang aman. Perbandingan performa pada penerapan kedua metode pengamanan data pada sistem komunikasi VoIP bertujuan untuk mengetahui protokol mana yang aman untuk diterapkan. Pada dasarnya ZRTP menggunakan pertukaran kunci Diffie-Hellman sebagai pertukaran kunci komunikasi antar client, yang mana kunci untuk komunikasi antar client tersebut menggunakan hash dari Diffie-Hellman dan dilakukan secara peer-to-peer melalui paket RTP VoIP. Sedangkan pada VPN IPsec terdapat protokol ESP (Encapsulating Security Payload) yang berfungsi sebagai enkripsi dan membuat tunnel transport RTP pada sistem komunikasi VoIP. Acuan standarisasi pengukuran kualitas VoIP yaitu rekomendasi dari ITU (International Telecomunication Union) yang merupakan standarisasi internasional untuk mengukur kualitas suara dalam komunikasi telephone. Pengujian dilakukan menggunakan variasi bandwidth 32, 64, 128, 256, 512, dan 1024 kbps. Parameter QoS (Quality of Service) yaitu delay, jitter, packet loss, dan throughput. Hasil pengujian QoS (Quality of Service) menunjukkan bahwa protokol VPN IPsec menghasilkan total delay rata-rata yang cukup rendah sebesar 69,78 % dibandingkan dengan VoIP ZRTP, throughputrata-rata yang lebih tinggi sebesar 15,53 % tetapi nilai jitter cukup tinggi dibadingkan dengan ZRTP. Sedangkan packet loss pada sistem komunikasi VoIP ZRTP menghasilkan nilai packet loss yang stabil yaitu 0 %, sedangkan untuk VoIP VPN IPsec menghasilkan packet loss mencapai 4,62 %.. MOS (Mean Opinion Score) menunjukan bahwa hasil perhitungan MOS Score yang menyatakan bahwa pemakaian limit bandwidth 128 kbps dan 256 kbps performa VoIP ZRTP lebih unggul sebesar 11,21 % dan 9,63 % dari performa VoIP VPN IPsec. Akan tetapi dalam pemakaian limit bandwidth 512 dan 1024 kbps performa VoIP VPN IPsec mulai mengalami peningkatan dan lebih unggul 1,18 % dan 3,23 % dari performa VoIP ZRTP.

scholarly journals Analisis Kinerja Video Conference pada Jaringan Mobile IPv6 Dengan Triangle Tunneling Method

EXPLORE ◽  
2019 ◽  
Vol 9 (2) ◽  
Author(s):  
Muhammad Multazam
Keyword(s):  
Packet Loss ◽  
Mobile Ip ◽  
Mobile Node ◽  
Mobile Ipv6 ◽  
Video Conference ◽  
Home Agent ◽  
Delay Jitter ◽  
Tunneling Method ◽  
Correspondent Node

Mobile IP merupakan protocol yang mendukung mobilitas dari user, dengan mobile IP sebuah node dapat terus berkomunikasi dengan node yang lain dimanapun dia berada. Mobile IPv6 memperbolehkan node yang telah mempunyai dukungan terhadap IPv6 untuk meninggalkan jaringan asalnya sementara node tersebut sedang memperbarui koneksinya ke internet. Hal ini berarti bahwa mobile IPv6 bisa mengidentifikasi tiap-tiap node dengan static address-nya, tanpa memperhatikan point of attachment nya ke internet.Mobile IPv6 yang digunakan pada makalah ini memakai metode triangle tunneling yaitu metode perutean pengiriman paket dimana ketika correspondent node (CN) mengirimkan paket menuju mobile node (MN), home agent (HA) mengambil alih paket tersebut dan mengirimkannya ke care of address dari MN dengan tunneling, sedangkan MN dapat mengirimkan langsung menuju CN. Pada makalah ini dilakukan pengujian video conference (multipont) pada jaringan Mobile IPv6 dengan metode triangle tunneling untuk diukur parameter-parameter QoS meliputi packet loss, delay, jitter, throughput dan MOS.Berdasarkan hasil uji coba dengan memperhatikan parameter-parameter QoS yang dianalisa terlihat bahwa kualitas MN yang berada di jaringan asalnya (homenet) lebih baik ketika berada di jaringan yang sedang dikunjungi (visitnet), hal ini disebabkan dengan metode triangle tunneling paket yang dikirim menuju MN dirutekan ke HA terlebih dahulu sehingga meningkatkan trafik jaringan, selain itu pada uji coba handover terdapat nilai rata-rata delay yang cukup besar yaitu 3.367 detik untuk paket video dan 3.269 detik untuk paket audio, karena pada saat handover, MN membutuhkan waktu untuk memperbaharui koneksi dan melakukan konfigurasi ulang jaringan yang ada.

scholarly journals Performance evaluation of VoIP technology in an extended service set, in concordance with IEEE 802.11g

2017 ◽  
Vol 15 (42) ◽  
pp. 85-100
Author(s):  
Marcelo David Núñez Cuadrado ◽  
Carlos Andres Jativa Huilcapi ◽  
Román Alcides Lara Cueva
Keyword(s):  
Packet Loss ◽  
The Other ◽  
Injection Technique ◽  
Conversational Analysis ◽  
Delay Jitter ◽  
Optimal Delay ◽  
Opinion Score ◽  
Service Degradation ◽  
Extended Service

In this paper, we evaluate the performance in function of the metrics associated to Quality of Service [QoS] and Quality of user Experience [QoE] in an experimental way in the VoIP service for G.711 and G.729 códecs. This was performed over an extended service set based on Wi-Fi technology in concordance with IEEE 802.11g standard using embedded systems. QoS related metrics are obtained by using the intrusive traffic injection technique. In addition, we assessed the QoE using the MOSc [Mean Opinion Score conversational] analysis. The best results were obtained for G.729, reaching up to 25 simultaneous injections with optimal delay, jitter and packet loss values according to the ITU-T recommendation for VoIP. However, the G.711 codec presented a better throughput. On the other hand, QoE evaluation indicates a slight superiority of G.729 in the MOSc appreciation. Finally, we conclude that packet loss and delay are the most influential metrics in VoIP service degradation.

scholarly journals Implementation of Hotspot Network for Internal Campus Communications Utilizing Smartphone and Free Software

2017 ◽  
Vol 1 (1) ◽  
pp. 33
Author(s):  
Pande Ketut Sudiarta ◽  
I Putu Ardana
Keyword(s):  
Packet Loss ◽  
Circuit Switching ◽  
Voice Communication ◽  
Campus Environment ◽  
Telephone Communication ◽  
Mobile Phone Technology ◽  
Limited Bandwidth ◽  
Telephone Exchange ◽  
User Data

VoIP (voice over internet protocol) telephone communication using a data network. There is a change in switching from circuit switching technology into packet switching. Phone exchange can now use the Personal Computer equipped VoIP applications. Even the development of mobile phone technology to make VoIP communication can be performed utilizing the Smartphone. VoIP applications commonly use such as WhatsApp, Line etc. However, this application will cut the user data packets and often the quality is not satisfactory due to limited bandwidth and location of the remote server. During this time, Udayana University campus at several locations has been equipped with Hotspot network is mostly used to connect to the Internet. Hotspot network can also be used for voice communications with VoIP technology by adding a VoIP server. This concept not raises communication costs and should produce sound quality will be better because of the close location of the server. Because that researchers need to develop a model of telephone communication network utilizing Smartphone hotspot and students to be able to communicate in a campus environment. Method of this research is to develop a network utilizing hotspots and VoIP telephone exchange using the mini PC installed software FreePBX. On the side of the Smartphone using the free soft phone application and for aircraft used FXS analog phone as a codec. Tests will be performed for the communication between Smartphone and Smartphone to FXS terms of QOS and MOS produced. The results obtained, if the latency and packet loss have a value corresponding to the Real Time Protocol (RTP), the obtained MOS appropriate for the codec used while with the same codec if the value of packet loss and latency results are high then MOS obtained be small or less good quality. So the quality of VoIP is highly dependent on the quality of the signal obtained hotspot. In general, VoIP communication using a Smartphone connected to the server on the network hotspot mini pc can be used as a voice communication on campus.

scholarly journals ANALISA QUALITY OF SERVICES (QoS) VOICE OVER INTERNET PROTOCOL (VoIP) DENGAN PROTOKOL H.323 DAN SESSION INITIAL PROTOCOL (SIP)

2012 ◽  
Vol 1 (2) ◽  
Author(s):  
Eko Budi Setiawan
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Internet Protocol ◽  
Mean Opinion Score ◽  
Quality Monitoring ◽  
Quality Of Services ◽  
Voice Over Internet Protocol ◽  
Delay Jitter ◽  
Opinion Score

Teknologi Voice over Internet Procotol (VoIP) tentu sangat menguntungkan bagi masyarakat luas karena dengan adanya VoIP maka sarana untuk melakukan komunikasi dari segi biaya yang dikeluarkan akan menjadi lebih sedikit apabila dibandingkan dengan media telepon konvensional biasa. Protokol H.323 dan Session Initiation Protocol (SIP) merupakan protokol yang dapat digunakan untuk membangun VoIP. Baik protokol H.323 dan SIP metodenya berbeda satu sama lain, sehingga tentunya perlu dilakukan suatu pengujian dan analisis untuk mengetahui kualitas yang dihasilkan dari VoIP dengan menggunakan kedua protokol tersebut. Parameter Quality of Service (QoS) yang digunakan untuk mengamati kualitas dari VoIP tersebut adalah delay, jitter, packet loss, serta Mean Opinion Score (MOS) yang diujikan pada 5 audience.Pengukuran QoS tersebut mengunakan tools VoIP Quality Monitoring VQManager yang dilakukan pada bandwidth sebesar 512 Kbps, 256 Kbps, 128 Kbps, dan 64 Kbps dengan Audio Codec G.711. Dari hasil pengujian yang dilakukan dengan skema percobaan yang sama, secara umum nilai dari parameter QoS yang didapat ketika menggunakan protokol H.323 bisa dikatakan lebih baik bila dibandingkan dengan protokol SIP. Kata Kunci :

Virtualized Fog Network with Load Balancing for IoT based Fog-to-Cloud

2020 ◽  
Vol 4 (3) ◽  
Author(s):  
Istabraq M. Al-Joboury ◽  
Emad H. Al-Hemiary
Keyword(s):  
Load Balancing ◽  
Packet Loss ◽  
Fog Computing ◽  
Transport Protocol ◽  
Network Simulator ◽  
Huge Number ◽  
Message Queue ◽  
Limited Bandwidth ◽  
Cloud Architecture ◽  
Fog Network

Fog Computing is a new concept made by Cisco to provide same functionalities of Cloud Computing but near to Things to enhance performance such as reduce delay and response time. Packet loss may occur on single Fog server over a huge number of messages from Things because of several factors like limited bandwidth and capacity of queues in server. In this paper, Internet of Things based Fog-to-Cloud architecture is proposed to solve the problem of packet loss on Fog server using Load Balancing and virtualization. The architecture consists of 5 layers, namely: Things, gateway, Fog, Cloud, and application. Fog layer is virtualized to specified number of Fog servers using Graphical Network Simulator-3 and VirtualBox on local physical server. Server Load Balancing router is configured to distribute the huge traffic in Weighted Round Robin technique using Message Queue Telemetry Transport protocol. Then, maximum message from Fog layer are selected and sent to Cloud layer and the rest of messages are deleted within 1 hour using our proposed Data-in-Motion technique for storage, processing, and monitoring of messages. Thus, improving the performance of the Fog layer for storage and processing of messages, as well as reducing the packet loss to half and increasing throughput to 4 times than using single Fog server.

ANALISIS DE CALIDAD DE SERVICIO EN REDES IPv6

2016 ◽  
Vol 26 (1) ◽  
pp. 7
Author(s):  
Jose Carlos Tavara Carbajal
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Traffic Control ◽  
Video Traffic ◽  
Control Software ◽  
Delay Jitter ◽  
End To End Delay ◽  
Analysis Of Results ◽  
Ipv6 Protocol

RESUMENEste documento tiene como objetivo analizar el comportamiento de la calidad del servicio del protocolo IPv6 sobre el tráfico de video, para esto se realizó sobre un entorno real y se llevó acabo el análisis de resultados a través de un software estadístico de control del tráfico.Palabras Clave.-  Calidad de Servicio, Ancho de Banda, Retardo, Fluctuación de Retardo, Pérdidas de Paquetes.ABSTRACTThis paper has aimed to analyze of the service quality of the IPv6 protocol on video traffic, this was about a real environment and was conducted analysis of results through statistical traffic control software. Key words- Quality of Service, Bandwidth, End to end delay, Jitter, Packet loss.

ANALISIS DE CALIDAD DE SERVICIO EN REDES IPv6

2016 ◽  
Vol 26 (1) ◽  
pp. 1
Author(s):  
Jose Carlos Tavara Carbajal
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Traffic Control ◽  
Video Traffic ◽  
Control Software ◽  
Delay Jitter ◽  
End To End Delay ◽  
Analysis Of Results ◽  
Ipv6 Protocol

Este documento tiene como objetivo analizar el comportamiento de la calidad del servicio del protocolo IPv6 sobre el tráfico de video, para esto se realizó sobre un entorno real y se llevó acabo el análisis de resultados a través de un software estadístico de control del tráfico.Palabras Clave.-  Calidad de Servicio, Ancho de Banda, Retardo, Fluctuación de Retardo, Pérdidas de Paquetes.ABSTRACT  This paper has aimed to analyze of the service quality of the IPv6 protocol on video traffic, this was about a real environment and was conducted analysis of results through statistical traffic control software.  Key words.- Quality of Service, Bandwidth, End to end delay, Jitter, Packet loss.

ANALISIS DE CALIDAD DE SERVICIO EN REDES IPv6

2016 ◽  
Vol 26 (1) ◽  
pp. 7
Author(s):  
Jose Carlos Tavara Carbajal
Keyword(s):  
Quality Of Service ◽  
Packet Loss ◽  
Traffic Control ◽  
Video Traffic ◽  
Control Software ◽  
Delay Jitter ◽  
End To End Delay ◽  
Analysis Of Results ◽  
Ipv6 Protocol

RESUMENEste documento tiene como objetivo analizar el comportamiento de la calidad del servicio del protocolo IPv6 sobre el tráfico de video, para esto se realizó sobre un entorno real y se llevó acabo el análisis de resultados a través de un software estadístico de control del tráfico.Palabras Clave.-  Calidad de Servicio, Ancho de Banda, Retardo, Fluctuación de Retardo, Pérdidas de Paquetes.ABSTRACTThis paper has aimed to analyze of the service quality of the IPv6 protocol on video traffic, this was about a real environment and was conducted analysis of results through statistical traffic control software. Keywords- Quality of Service, Bandwidth, End to end delay, Jitter, Packet loss.

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