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Prasaja Wikanta
"Communication with VoIP can encourage us to change the old network of point to point dedicated line to more cheap and efficient network, such as MPLS. VOIP is more economical in bandwidth usage, with not so much different in quality. Conversion of analog a udio signal to its digital counterpart are done by Cisco Voice router 2811, and forward it to IP network. VoIP can be extended to serve conference. So communication is not just for 2 participants, but able to serve 3 or more participants.
We developed VoIP system that can accommodate one or more participants to join conversation existed, thus forming a conference. We use conference bridge which is shipped together with Asterisk application, to multiplex all voice signal. In order to keep load of server Asterisk low, we use distributed server model. Furthermore, centralized conference server will increase bandwidth used in that server, hence more cost for operation.
Redundancy and QoS also included in our design considerations. We also developed The Conference Terminal that able to choose which conversation he want to join with signal DTMF.

Communication VoIP peut nous encourager de remplacer traditionnel ligne dédié point à point avec nouvel reseau qui est moins cher et plus efficace, tels que MPLS. VoIP est plus économisé dans l'utilisation de débit que line dédié, avec la qualité n'est pas trop différent. Nous utilisons Cisco routeur voix 2811 de convertir signal audio analogue à numérique, et puis les transmet sur réseau IP. VoIP peut servir la conférence voix. Alors la conversation n'est pas juste entre 2 parties, mais il est possible de 3 parties ou plus.
Nous développons système VoIP qui peut accueillir un ou plus parties de joindre la conversation entre 2 parties existe. Il établie alors, la conférence. Nous utilisons pont de la conférence qui est servi par Asterisk. Afin de garder la charge de serveur n'est pas excessive, j'applique la méthode serveur distribue. De plus, serveur de conférence centralise va augmenter le débit utile qui font le prix de connexion plus cher.
Redondant et QoS sont également comptée dans notre conception. Nous développons le Terminal Conférence qui peut choisir quel conversation il veut le joindre en utilisant signal DTMF.
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Depok: Fakultas Teknik Universitas Indonesia, 2011
T29584
UI - Tesis Open  Universitas Indonesia Library
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Chakraborty, Tamal
"This book offers an accessible introduction and practical guide to Voice over Internet Protocol (VoIP) technology, providing readers with the know-how to solve the problems encountered in applying VoIP technology across all types of network. It incorporates the latest research findings and brings readers up to date with the challenges that are faced by researchers developing novel applications of VoIP.
The authors discuss the general architecture of VoIP technology, along with its application and relevance in conventional and emerging wireless communication networks, including Wireless Local Area Networks (WLANs), Worldwide Interoperability for Microwave Access (WiMAX), Long Term Evolution (LTE) and Cognitive Radio Networks. The book also includes Quality of service (QoS) studies under dynamic and unpredictable network conditions, which examine the reliability of both legacy systems And the upcoming pervasive computing systems. Further, it explains how the heuristic-based learning algorithms that are used in VoIP communications may help develop today’s technology in the area of autonomous systems.
This book is a valuable source of information for academics and researchers, as it provides state-of-theart research in VoIP technology. It is also of interest to network designers, application architects, and service providers looking for a coherent understanding of VoIP across a wide range of devices, network applications and user categories."
Switzerland: Springer Cham, 2019
e20502223
eBooks  Universitas Indonesia Library