Voice Over Internet Protocol

Author(s):  
Indranil Bose ◽  
Fong Man Chun

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).

2021 ◽  
Vol 17 (1) ◽  
pp. 11-22
Author(s):  
Wahyu Adi Prijono

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.


Author(s):  
Priya Chandran ◽  
Chelpa Lingam

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.


2012 ◽  
Vol 1 (1) ◽  
Author(s):  
Domiko Fahdi Jaya Patih

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


Author(s):  
Indranil Bose

Today, Internet technologies have pervaded every corner of our society. More and more people are benefiting from the Internet in one way or the other. One of the current Internet technologies that may benefit us greatly is voice over Internet protocol (VoIP). According to Hardy (2003, p. 2), VoIP is “the interactive voice exchange capability carried over packet-switched transport employing the Internet protocol.” With VoIP technology, one can call anyone in this world at a lower cost, compared to traditional telephone systems. However, VoIP technology has one significant drawback. It has a low degree of reliability. From experimental results it is known that VoIP can achieve only 98% reliability. The service down time per year for VoIP is almost 20 working days (175 hours). For most companies and government organizations, such a degree of reliability is unacceptable since the traditional telephone system can achieve 99.999% reliability with a service down time of only five minutes per year (Kos, Klepec, & Tomaxic, 2005). As a result, quality of service (QoS) is an important concept for VoIP. Using QoS, VoIP may be able to overcome its limitation in reliability. QoS is often defined as the capability to provide resource assurance and service differentiation in a network. The definition includes two important terms—resource assurance and service differentiation. Resource assurance provides a guarantee about the amount of network resources requested by the user. On the other hand, service differentiation provides higher priority of getting network resources to those applications that have critical latency constraints. Given the importance of low latency for voice communication, it is not difficult to predict that QoS will assume greater importance in the VoIP industry as this technology gains popularity in the mass market. It is reported that VoIP is aggressively growing, and this growth is expected to continue in the coming years.


2019 ◽  
Vol 5 (1) ◽  
pp. 55-64
Author(s):  
Ardi Windiarto ◽  
Kholilatul Wardani

Makalah ini membahas desain layanan jaringan komunikasi VoIP Server menggunakan Raspberry Pi sebagai alat komunikasi wireless. VoIP server berbasis Raspberry Pi menggunakan sistem operasi RasPBX. Di dalam sistem operasi RasPBX sudah ada software asterisk yang berfungsi sebagai softswicth. Client VoIP menggunakan zoiper sebagai softphone. Alat ini dilengkapi dengan fitur GSM gateway yaitu fitur yang dapat menghubungkan jaringan VoIP ke jaringan GSM. Fitur GSM gateway ini menggunakan modem GSM sebagai jembatan yang menghubungkan jaringan VoIP dengan jaringan GSM. Persentase keberhasilan panggilan VoIP ke VoIP, VoIP ke GSM, dan GSM ke VoIP mencapai 100%. Berdasarkan hasil pengujian Quality of services (QoS) pada panggilan VoIP ke GSM, dihasilkan rata-rata delay sebesar 12,11 ms yang termasuk dalam kategori kualitas baik, Troughput sebesar 0,151, jitter sebesar 0,052 ms yang termasuk dalam kategori kualitas baik, dan packet loss sebesar 0% yang termasuk dalam kategori kualitas sangat baik. Jangkauan maksimal antara client VoIP ke server agar komunikasi berjalan dengan baik adalah 100 meter dalam kondisi Line Of Sight (LOS). Pengujian dengan jarak 25 m dalam kondisi Non Line Of Sight (NLOS), masih menghasilkan komunikasi yang baik. Berdasarkan hasil pengujian kuisioner dari 30 pengguna, dihasilkan nilai MOS 3,88 yang termasuk dalam kategori kualitas cukup baik.


Author(s):  
Md. Anwar Hossain ◽  
Mst. Sharmin Akter

Routing is a design way to pass the data packet. User is assigns the path in a routing configuration. A significant role played by the router for providing the dynamic routing in the network. Structure and Configuration are different for each routing protocols. Next generation internet protocol IPv6 which provides large address space, simple header format. It is mainly effective and efficient routing. It is also ensure good quality of service and also provide security. Routing protocol (OSPFv3) in IPv6 network has been studied and implemented using ‘cisco packet tracer’. ‘Ping’ the ping command is used to check the results. The small virtual network created in Cisco platform .It is also used to test the OSPFv3 protocol in the IPv6 network. This paper also contains step by step configuration and explanation in assigning of IPv6 address in routers and end devices. The receiving and sending the packet of data in a network is the responsibility of the internet protocol layer. It also contains the data analysis of packet forwarding through IPv6 on OSPFv3 in simulation mode of cisco packet virtual environment to make the decision eventually secure and faster protocol in IPv6 environment.


bit-Tech ◽  
2018 ◽  
Vol 1 (1) ◽  
pp. 1-8
Author(s):  
Riki Riki ◽  
Aditiya Hermawan ◽  
Yusuf Kurnia

TCP\IP protocol can be connected to various computer data networks in the world. This protocol increasingly exists and is needed so that many parties develop it to vote through this protocol. Voice Over Internet Protocol technology is the answer to that desire. This technology is able to convert analog voice (human voice) into data packets then through public internet data networks and private intranet data packets are passed, so that communication can occur. With VoIP communication costs can be reduced so that it can reduce investment costs and conversations (cost saving) or even up to 100% free. VoIP implementation can be done by designing a wireless VoIP network (cable) using 3CXSystemPhone software as a PBX. In this scientific work the software used is 3CXSystemPhone 11.0, where SIP is a VoIP server which is a freeware software, in its application only requires one PC server and several PC clients (2 for example) that are connected to each other


Author(s):  
Thomas M. Chen

The founding of the Bell Telephone System, the public switched telephone network (PSTN), has evolved into a highly successful global telecommunications system. It is designed specifically for voice communications, and provides a high quality of service and ease of use. It is supported by sophisticated operations systems that ensure extremely high dependability and availability. Over the past 100 years, it has been a showcase for communications engineering and led to groundbreaking new technologies (e.g., transistors, fiber optics). Yet it is remarkable that many public carriers see their future in Internet protocol (IP) networks, namely the Internet. Of course, the Internet has also been highly successful, coinciding with the proliferation of personal computers. It has become ubiquitous for data applications such as the World Wide Web, e-mail, and peer-to-peer file sharing. While it is not surprising that the Internet is the future for data services, even voice services are transitioning to voice over Internet protocol (VoIP). This phenomenon bears closer examination, as a prime example explaining the success of the Internet as a universal communications platform. This chapter gives a historical development of the Internet and an overview of technical and nontechnical reasons for the convergence of services.


Techno Com ◽  
2020 ◽  
Vol 19 (1) ◽  
pp. 1-11
Author(s):  
Agus Heriyanto ◽  
Lailis Syafaah ◽  
Amrul Faruq

Di dalam komunikasi Voice over Internet Protocol (VoIP) mengenal beberapa macam protocol tambahan selain protocol standar internet Transfer Control Protocol/Internet Protocol (TCP/IP), beberapa diantaranya adalah protocol Session Initation Protocol (SIP), Inter-Asterisk eXchange (IAX) dan H.323. Performansi perlu dijaga mengingat VoIP mempunyai kemungkinan melakukan berbagai cara kompresi untuk menciptakan efisiensi saluran dan pemilihan protocol yang tepat. Teknologi VoIP pada dasarnya tidak memiliki jaminan keamanan pada setiap komunikasi. Keamanan ketika melakukan komunikasi suara merupakan sesuatu yang sangat penting karena menyangkut privasi penggunanya. Penggunaan Virtual Private Network (VPN) merupakan salah satu solusi untuk menutup celah keamanan pada kasus di atas. Analisis yang dilakukan pada artikel ini adalah performa yang dihasilkan VoIP yang menggunakan protocol IAX dan SIP. Penelitian ini mengahasilkan kesimpulan bahwa performansi yang paling baik digunakan untuk membangun sistem komunikasi VoIP adalah protocol IAX dengan menggunakan sistem keamanan VPN Point to Point Protocol (PPTP) dikarenakan nilai Quality of Service (QoS)  lebih tinggi daripada protocol SIP dan juga terbukti lebih aman saat diterapkan sistem keamanan Virtual Private Network Point to Point Protocol (VPN PPTP).


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