PENGARUH PENGGUNAAN CODEC STANDART ITU G.729 TERHADAP SISTEM KOMUNIKASI VOIP

2021 ◽  
Vol 17 (1) ◽  
pp. 11-22
Author(s):  
Wahyu Adi Prijono
Keyword(s):  
Communication System ◽  
Communication Systems ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Data Communications ◽  
Voice Communication ◽  
Internet Telephony ◽  
Voice Signal ◽  
Ip Network ◽  
Packet Switch

Voice over Internet Protocol (VoIP) is a technology that is capable of passing voice traffic, in the form of packets through the network Internet Protocol (IP). IP network itself is a data communications network based packet-switch. The voice signal before experiencing bundled voice coding or format conversion of sound into digital form that can be passed over an IP network. Telephony, Internet telephony, or termed VoIP (Voice Over Internet Protocol.This communication system use VoIP (Voice over Internet Protocol), ie voice calls over data services (internet). This communication was developed using Android-based devices. Based on characteristics, android devices are open source, so users do not need to have a license to be able to have android-based devices. In addition, the android device that must be connected to a SIP (Session Iniation Protocol) is a data service that can be done with a paid subscription of the user of the operator using a conventional pulse. Telecommunications designed will use a hybrid system, the merger between VoIP communications with data communications GSM network. With basic calculations where Coding standards G 729, is a standard that can be used for voice communication system through data networks with rate of 8 Kbps. The implementation of the G729 codec is effect on communication systems VOIP.

Voice Over Internet Protocol

2008 ◽  
pp. 663-669 ◽  
Author(s):  
Indranil Bose ◽  
Fong Man Chun
Keyword(s):  
Internet Protocol ◽  
Data Networks ◽  
The Internet ◽  
Voice Over Internet Protocol ◽  
Voice Communication ◽  
Voice Transmission ◽  
Voice Signal ◽  
Public Networks ◽  
Unified Messaging

One of the hottest technologies these days is voice communication over packet-switched data networks. This is known as voice over Internet protocol (VoIP). Hardy (2003) defines VoIP as “the interactive voice exchange capability carried over packet switched transport employing the Internet protocol.” With VoIP, voice signal from the sender is digitized into packets that are transmitted to the receiver through a network,which often includes the Internet. As the Internet is a free resource, the communication cost of VoIP is much lower than that of traditional telephone systems. This is a major advantage of VoIP. VoIP system also increases the efficiency and service quality of businesses. As a result of VoIP, many advanced applications can be built and these include unified messaging, video conferencing, and ring list. But VoIP is not without its limitations. Its main drawback is low reliability. It also suffers from uncertain quality of voice transmission. In addition, it cannot guarantee security because it uses public networks. Although the idea of VoIP was known from the 1970s, it did not become commercially viable until 1995, when Vocaltec became the first company to produce the first commercially available VoIP product (Varshney et al., 2002).

Comparison of QoS Performance Over WLAN, VoIP4 and VoIP6

2015 ◽  
Vol 2 (11) ◽  
pp. 42
Author(s):  
Esra Musbah Mohammed Musbah ◽  
Khalid Hamed Bilal ◽  
Amin Babiker A. Nabi Mustafa
Keyword(s):  
Performance Analysis ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Ip Network ◽  
Qos Parameters ◽  
Software Program ◽  
Jitter Delay ◽  
Qos Performance ◽  
Delay Variation ◽  
A Performance

VoIP stands for voice over internet protocol. It is one of the most widely used technologies. It enables users to send and transmit media over IP network. The transition from IPv4 to IPv6 provides many benefits for internet IPv6 is more efficient than IPv4. This paper presents a performance analysis of VoIP over WLAN using IPv4 and IPv6 and OPNET software program to simulate the protocols and to investigate the QoS parameters such as jitter, delay variation, packet send, and packet received and throughputs for IP4 and IP6 and compare between them.

scholarly journals BPL-PLC Voice Communication System for the Oil and Mining Industry

Energies ◽  
2020 ◽  
Vol 13 (18) ◽  
pp. 4763
Author(s):  
Grzegorz Debita ◽  
Przemysław Falkowski-Gilski ◽  
Marcin Habrych ◽  
Grzegorz Wiśniewski ◽  
Bogdan Miedziński ◽  
...  
Keyword(s):  
Fiber Optics ◽  
Communication System ◽  
Communication Systems ◽  
High Efficiency ◽  
Coal Dust ◽  
Mining Industry ◽  
Power Line ◽  
Communication Link ◽  
Voice Communication ◽  
Medium Voltage

Application of a high-efficiency voice communication systems based on broadband over power line-power line communication (BPL-PLC) technology in medium voltage networks, including hazardous areas (like the oil and mining industry), as a redundant mean of wired communication (apart from traditional fiber optics and electrical wires) can be beneficial. Due to the possibility of utilizing existing electrical infrastructure, it can significantly reduce deployment costs. Additionally, it can be applied under difficult conditions, thanks to battery-powered devices. During an emergency situation (e.g., after coal dust explosion), the medium voltage cables are resistant to mechanical damage, providing a potentially life-saving communication link between the supervisor, rescue team, paramedics, and the trapped personnel. The assessment of such a system requires a comprehensive and accurate examination, including a number of factors. Therefore, various models were tested, considering: different transmission paths and types of coupling (inductive and capacitive), as well as various lengths of transmitted data packets. Next, a subjective quality evaluation study was carried out, considering speech signals from a number of languages (English, German, and Polish). Based on the obtained results, including both simulations and measurements, appropriate practical conclusions were formulated. Results confirmed the applicability of BPL-PLC technology as an efficient voice communication system for the oil and mining industry.

scholarly journals Implementasi Trunk Interkoneksi Multi Server menggunakan Singleboard Komputer

2021 ◽  
Vol 11 (2) ◽  
pp. 96-100
Author(s):  
Alwalid Nouvatie ◽  
Martono Dwi Atmadja ◽  
Waluyo Waluyo
Keyword(s):  
Packet Loss ◽  
Internet Protocol ◽  
Voice Over Internet Protocol ◽  
Ip Telephony ◽  
Ip Address ◽  
Single Board Computer ◽  
Internet Telephony ◽  
Private Branch Exchange ◽  
Ip Phone ◽  
Multi Server

Voice over Internet Protocol (juga disebut VoIP, IP Telephony, Internet telephony atau Digital Phone) adalah teknologi yang memungkinkan percakapan suara jarak jauh melalui media internet. Data suara diubah menjadi kode digital dan dialirkan melalui jaringan yang mengirimkan paket-paket data, dan bukan lewat sirkuit analog telepon biasa. Dalam komunikasi VoIP, pemakai melakukan hubungan telepon melalui terminal yang berupa PC atau telepon biasa. Dengan bertelepon menggunakan VoIP, banyak keuntungan yang dapat diambil diantaranya adalah dari segi biaya jelas lebih murah dari tarif telepon tradisional, karena jaringan IP bersifat global. IP Phone dapat di tambah, dipindah dan di ubah. Hal ini karena VoIP dapat dipasang di sembarang ethernet dan IP address, tidak seperti telepon konvensional yang harus mempunyai port tersendiri di Sentral atau PBX (Private branch exchange). Dalam penelitian ini mengimplementasikan keterhubungan antar server menggunakan single board computer yang di install sistem operasi Elastix yang bertujuan untuk mengimplementasikan prefix untuk antar server dan menggunakan beberapa codec audio. Hasil penelitian telepon menggunakan prefix dan tanpa prefix sebanyak 6 client atau 3 pasng panggilan secara bersamaan nilai packet loss tertinggi pada codec speex dengan prefix sebesar 2,34%. Nilai bandwidth  tertinggi yang digunakan adalah dengan prefix codec PCMU dengan rata-rata 82,3 Kbps dan tanpa prefix 79,3 Kbps. Kata kunci :   Server, VoIP, IP Telphony, Internet telephony, Digital Phone, IP Address, PBX, Codec, Prefix.

scholarly journals A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication

2018 ◽  
Vol 8 (5) ◽  
pp. 2926
Author(s):  
Priya Chandran ◽  
Chelpa Lingam
Keyword(s):  
Packet Loss ◽  
Loss Rate ◽  
Voice Quality ◽  
Internet Protocol ◽  
Packet Loss Rate ◽  
Voice Over Internet Protocol ◽  
Network Delay ◽  
Voice Communication ◽  
Voice Packets

Factors like network delay, latency and bandwidth significantly affect the quality of communication using Voice over Internet Protocol. The use of jitter buffer at the receiving end compensates the effect of varying network delay up to some extent. But the extra buffer delay given for each packet plays a major role in playing late packets and thereby improving voice quality. As the buffer delay increases packet loss rate decreases, which in general is a very good sign. However, an increase of buffer delay beyond a certain limit affects the interactive quality of voice communication. In this paper, we propose a statistical framework for adaptive playout scheduling of voice packets based on network statistics, packet loss rate and availability of packets in the buffer. Experimental results show that the proposed model allocates optimal buffer delay with the lowest packet loss rate when compared with other algorithms.

scholarly journals Rancang Bangun Perangkat Lunak Billing dan Implementasi Voice Over Internet Protocol

2013 ◽  
Vol 4 (2) ◽  
pp. 603
Author(s):  
Honni Honni
Keyword(s):  
Data Collection ◽  
Communication System ◽  
Internet Protocol ◽  
Session Initiation Protocol ◽  
Voice Over Internet Protocol ◽  
Design Development ◽  
Data Network ◽  
Signaling Protocol ◽  
Telephone Communication ◽  
Analysis Design

The rapidly evolving communication system enables applications for telephone communication to be carried over the data network known as VoIP (voice over internet protocol). SIP (session initiation protocol) as the signaling protocol is text-based VoIP which can be implemented easily in comparison with other signalingprotocols. The purpose of this paper is designing and implementing VoIP billing up to the company to provide additional facilities for enterprise customers. The methods start with data collection, analysis, design, development, and implementation. The result achieved is a system of VoIP with SIP and Asterisk software which has functions of PBX to provide additional facilities such as VoIP which is a plus for the company and customers. After implemented, the VoIP system and billing features are found work well.

scholarly journals ANALISA PERANCANGAN SERVER VOIP (VOICE INTERNET PROTOCOL) DENGAN OPENSOURCE ASTERISK DAN VPN (VIRTUAL PRIVATE NETWORK) SEBAGAI PENGAMAN JARINGAN ANTAR CLIENT

2012 ◽  
Vol 1 (1) ◽  
Author(s):  
Domiko Fahdi Jaya Patih
Keyword(s):  
Internet Protocol ◽  
Virtual Private Network ◽  
The Internet ◽  
Voice Over Internet Protocol ◽  
Technology Infrastructure ◽  
Voice Communication ◽  
Telecommunication Technology ◽  
Communication Needs ◽  
Private Network ◽  
Internet Network

Abstrack Voice over Internet Protocol (VoIP) is a technology that utilizes the Internet Protocol to provide real-time voice communication. VoIP technology is a today telecommunication technology, where the costs of the technology infrastructure is much cheaper than the telecommunications technology that is commonly used today. Forming elements are VoIP User Agent, Proxy, Protocol and Coder-Decoder (CODEC). Asterisk is a softswicth to operate a proxy, which is based on session initiation protocol (SIP). 10.10 Ubuntu operating system as a VoIP server is flexible to support a package of performance Asterisk. The goal of this research is to build Asterisk-based VoIP server, that can be developed in further research as needed. The methodology of research conducted, is devided by two, the study of literature and experimental. The research was conducted at the installation that has been built before the Internet network. VoIP so here functioned as maximizing existing internet network is to reduce expenses communication needs. Services provided in this study form with voice and video call services client to server, client to client call, video call conferencing, video conferencingKey Word : Voice over Internet Protocol (VoIP), Asterisk, Session Initiaton Protocol (SIP), VPN

scholarly journals Distributed Coding for OFDM based Cooperative HF Radio Communication System

2019 ◽  
Vol 8 (2) ◽  
pp. 3138-3142
Keyword(s):  
Communication System ◽  
Channel Coding ◽  
Communication Systems ◽  
Future Generation ◽  
Radio Communication ◽  
Voice Communication ◽  
Radio System ◽  
Distributed Coding ◽  
High Data ◽  
Radio Communication System

High frequency (HF) radio system now-a-days used to establish the communication in an area which is isolated from the outside world due to natural calamities. Conventional HF systems are associated with analog voice communication systems; is now shifted to digital voice communication to meet the demands as expected for high data rates for transmission. Cooperative communication is different from conventional relay-assisted HF systems and aims to support the challenging expectation of future generation HF communication systems. Recent days distributed coding is a variety of channel coding developed in a distributed manner for cooperative wireless networks. In this paper we present an overview of various distributed coding design in OFDM based cooperative HF radio communication system.

Design of Wireless Voice Communication System in Underground Coal Mine Based on ZigBee

2014 ◽  
Vol 548-549 ◽  
pp. 1402-1406 ◽  
Author(s):  
Da Hu Wang ◽  
Qie Qie Zhang ◽  
Yi Fan Sun
Keyword(s):  
Wireless Communication ◽  
Communication System ◽  
Communication Systems ◽  
Ieee 802.15.4 ◽  
High Reliability ◽  
Voice Communication ◽  
Communication Device ◽  
Communication Devices ◽  
Voice Message ◽  
Voice Data

For disadvantages of the present mine voice communication systems, a kind of wireless voice communication system based on Zig Bee is put forward. The paper provides detailed informations about hardware and software of the wireless voice communication device. In the system, adopt CC2530 as RF sending-receiving unit of voice communication node, convert speech signals to digital or analog signals by CSP1027, encode or decode quantized voice data by AMBE voice codec technology and realize voice message two-way wireless communication over the Zig Bee wireless communication protocol IEEE 802.15.4 between voice communication devices. Experiments have shown that voice communication device can get a clear voice and have a high reliability in the effective distance of communication, meet the requirements of voice communication.

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