Voice over IP

In satellite networks, voice over IP performance is degraded by long delays and low bandwidth. Both call setup time and quality of service (QoS) for voice calls are affected. Separate studies have been undertaken for these performance metrics. In this paper, we have carried out experiments using different voice codecs to evaluate SIP call setup time and QoS parameters together.

We contribute to the study of the security of voice in IP (Internet Protocol) network, which will become in the near future, a universal standard of voice and video networks Telecommunications. As with any phone call, it is a need to encrypt communication to respect the rights and privacy of each person. We implement the security of voice in IP packets and study material resource consumption on the establishment of this system. This is the major problem with this kind of technology that is currently experiencing various attacks threatening all communication systems.

Voice over IP (VoIP) introduces new ways of communication, while utilizing existing data networks to provide inexpensive voice communications worldwide as a promising alternative to the traditional PSTN telephony. SPam over Internet Telephony (SPIT) is one potential source of future annoyance in VoIP. A common way to launch a SPIT attack is the use of an automated procedure (bot), which

Since the liberalization of the telecommunication sector in Uganda in 1998, the key services have been voice, data and low rate Internet. The backbone link to the rest of the world has been via satellite which is characterized by low bandwidth limiting broadband usage in Uganda. With the arrival of undersea cables at the East African coast, Uganda is now linked to the rest of the world via optic fiber which offers high bandwidth rendering access to broadband services. Internally, wired, Optic fiber and wireless technologies such as 3G mobile and WiMAX infrastructure have brought broadband services within reach of the urban population. These improvements have revolutionized telecommunication by making broadband access more affordable and enabling converged voice, data and video/entertainment services to the urban population.  Rural areas with low level or absent grid power and telecommunication infrastructure to deliver such services are still in a broadband blackout. In this paper, we discuss the challenges to rural broadband deployment in Uganda and propose possible technical solutions to address these challenges.

Currently two standards exist for signaling and control of voice over IP calls, namely ITU-T Recommendation H.323 and the IETF Session Initiation Protocol (SIP). Although there are a significant number of similarities between these two protocols, they behave significantly different when providing different higher level services. On one hand, SIP provides flexibility with broader scope, offering functions specifically designed to

The paper summarizes and proofs that existing VoIP programs are powerful and flexible enough to build complex applications. The applications are independent and can be used either stand-alone or within carrier-grade Telco environments. The described example of building a call center application (outbound calling) shows the power that lies not only in existing open-source Session Initiation Protocol (SIP) servers, but also in simple SIP tools. The costs for implementing those systems are at a very low level, as the functions each project performs are rather kept simple, but the combination possibilities are tremendously flexibility. The presented application has been deployed successfully within NGNlab as a Proof of Concept (PoC).

Voice over IP (VoIP), a modern form of telecommunications, requires real-time transmission.  However, there are limitations (e.g., packet loss and delay) which result in degradation in quality of voice transmission over an IP network.  ITU-T, a sector of the International Telecommunication Union, has issued recommendations for VoIP quality based on a Mean Opinion Score (MOS) derived primarily from studies on European languages.  ITU-T has acknowledged that there is an issue related to dependence on language/culture/nationality and the Quality of Experience (QoE) of multimedia, including MOS.  Because ITU-T standards have yet to be adopted for measurement of voice quality for tonal languages (e.g. Thai), the research in this thesis has several aims as follows. First, it is to carry out detailed assessment of subjective voice quality for the standard spoken Thai language and for native Thai speakers in a Thai environment.  Second, to compare voice quality perceived by Thai users of four VoIP codecs, G.729, G.711A-law, G.722 and G.723.1 (at 5.3 kbps). It has been found that Thai users found no significant difference between three of the codecs with a slight preference for G.722 over G.711A-law over G.729 codec. However, they ranked G.723.1 as having significantly poorer voice quality than the others. Therefore G.729 is recommended as the best choice because it has MOS in the range 4.13- 4.18 but requires a lower bandwidth than G.722 or G.711A-law. Third, it is to propose acceptable voice quality standards for VoIP services for Thai language and for native speakers of Thai.  It has been found that, on average, Thai users expected VoIP quality equivalent to a MOS of 3.41, i.e., fair quality. Finally, this thesis proposes two new models for Thai languages and Thai users, namely a Thai subjective-VoIP Quality Evaluation (ThaiVQE) model and an Enhanced E-model  (E2-model), an objective method. The ThaiVQE model focused on two major network parameters, packet loss and packet delay, and a G.711A-law codec. The E2-model was obtained by including a subjective Thai bias factor in a standard E-model.  The results showed that both models gave significant accuracy and reliability improvements compared to the standard E-model, with error reduction more than 20%. Therefore, both of the new models can support voice quality measurement in Thai environments with high accuracy, reliability and confidence.
(Total 169 pages) 

Keywords  :  VoIP, MOS, E-model, packet loss, packet delay, codec, Thai

Depuis les années 1990, Internet a été utilisé afin de diminuer les coûts des communications téléphoniques. Les communications ainsi établies coûtent seulement le prix de deux communications locales (une communication locale à chaque extrémité).

Toutefois, les enjeux de la voix sur IP sont aussi techniques et dépassent la simple idée de la communication téléphonique à moindre coût. Du fait de la convergence voix, données et images, il devient plus facile de gérer un support de transmission unique pour l'ensemble des services (tout sur IP).

SDN (Software Defined Networking) is a revolutionary technology
that is currently indeed in demand in various network services. VoIP
services can be greatly improved with SDN being used in their network.
In this lab, we review the technologies we are going to use in our
experiment and go through the background needed in our lab. Then,
we show how we built the lab itself in its different phases. Lastly, we
present the results we got from this lab. However, that doesn’t summarize
the capabilities of any of the used technologies but, it rather
gives a glimpse of what can be done with them especially SDN.

"Of all the AVT techniques, voice-over has probably been the least studied and the least valued.
This state of affairs can be found in not only western Europe (where voice-over is limited to
mostly non-fiction programmes) but also in Poland (where voice-over remains the most widely
applied technique of translation for feature films for the television and DVD markets. The aim of
this contribution is to re-evaluate some of the existing academic prejudices against voice-over
and to highlight its advantages in comparison with subtitling and dubbing, given that voice-over
is free of some specific constraints that are present in the other two AVT techniques. The
analysis ± illustrated by selected examples taken from the TV science-fiction series Star Trek ±
focuses on the interaction between the key factors in successful voice-over: (1) the acoustic
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quality and the quantity of translated text and (3) the timbre and LQWRQDWLRQRIWKHUHDGHU¶VYRLFH
and (4) the way in which the reader synchronises the reading with the original soundtrack. In the
conclusion, the author proposes that the voice-over of feature films could be improved
dramatically by transforming it into DµYRLFH-in-EHWZHHQ¶WHFKQLTXH"

In this paper we discuss how OPNET can be applied in advanced networking education, enhancing the practical skills of future network engineers. At the KU Leuven Master students in Electrical Engineering focusing on Telecommunications have to ...

Concept of call center technology with interactive voice response as one of main options on company to improve service to the customer. This technology supports implementation of CRM concepts, that able to maintain customer for keeping a loyalty to the company. This service is correspond with customer. Costumer is an object that will decide a future business development. Company will eventually lose market share if they do not notice customer as its object. However, the problems occurred for implementation of development service cost is quite large if we buy ready-made product in software house. Some companies choose to focus on IT development as cost center then it needs to develop an IVR call center service based VoIP. Based on the need to develop a call center IVR services with advanced VoIP-based access via PSTN, CDMA and GSM. The results of a study is the comparison of the use of VoIP-based IVR systems with commercial IVR system.

Voice over Internet Protocol (VoIP) is developed for voice communications system based on voice packets transmitted over IP network with real-time communications of voice across networks using the Internet protocols. Quality of Service (QoS) mechanism is applied to guarantee successful voice packets transmitted over IP network with reduced delay or drop according to assigned priority of voice packets. In this paper, the goal of simulation models is present to investigate the performance of VoIP buffer size for improving quality of service (QoS) through buffer optimization with the simulation results by using OPNET modeler version 14.5. The performance of the proposed algorithm is analyzed and compared the quality of service for VoIP. The final simulated result shows that the VoIP service performance best buffer size 128 Kb over WiMAX network.

El principal interés de 5G está en su capacidad para soportar una nueva ola de servicios, más allá de la voz, destacando las aplicaciones de IoT (telemedida, telecontrol, tecnologías para vestir, ciudades inteligentes, casas inteligentes, coche conectado, etc.). Sin embargo, la voz tiene un valor muy importante para conseguir que el ser humano interactúe de forma sencilla con todos estos dispositivos y aplicaciones, teniendo las operadoras la oportunidad de ofrecer servicios avanzados de voz seguros y fiables con capacidades de inteligencia artificial. En este artículo explican aquí los detalles tecnológicos para ofrecer vo5G y sus principales beneficios.

— As with any other technological advancement in use in today's age, security threats are proving to be the major challenges and risks. Knowledge about these security vulnerabilities presents an avenue of protecting organizational assets against virtual attacks. VoIP phone systems are becoming increasingly popular in today's society for business and personal purposes. VoIP services are increasingly productive and cheap, thereby, providing adopters a competitive edge. Voice over Internet Protocol (VoIP) is a widely deployed service since the commencement of voice and data integration. This was done in a bid to reduce cost and management concerns. VoIP uses the same infrastructure as traditional data networks and thus, inherits all the security challenges of a data network. In addition, VoIP exhibit self-inflected problems resulting from network components and the protocol adopted. This paper present the security threats witnessed in VoIP telecommunication. First, the paper explores a brief overview of VoIP techniques including network components, structure, standards and protocols, data processing techniques and quality of service. Second, the paper discusses the security threats in tandem with confidentiality, integrity and availability principle. Examples of security issues under consideration include; spamming, identity spoofing, call tempering, DoS, and Man-in-the-middle attacks among others. Finally, the paper will outline the common countermeasures adopted to mitigate the threats.