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ANALISA QOS PADA CODEQ G711 DALAM JARINGAN VOIP BERBASIS PROTOKOL SIP

Darmawan, Adi (2011) ANALISA QOS PADA CODEQ G711 DALAM JARINGAN VOIP BERBASIS PROTOKOL SIP. Other thesis, University of Muhammadiyah Malang.

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Abstract

The development of human needs, especially in the field of communications has created a voice communication services such as VoIP. Economically VoIP has become an alternative service in addition to the PSTN with a relatively cheap rate. The basic concept in the provision of services such as VoIP telephony is a telephony service should be able to provide a minimum level of services already provided by the PSTN with various completeness of its features. As a reference, the VoIP is supported by a variety of supporting technologies such as SIP, and codecs. SIP role in a variety of communication sessions, while the codec plays a role in changing the voice signals into digital signals that can be transmitted in IP networks. Satisfaction in the use of VoIP services is known as QoS. QoS becomes a reference in user satisfaction in using the services which is contained several parameters such as throughput, packet loss, delay and jitter. VoIP services in a LAN network using the SIP protocol and G711 codecs. This system was tested with a simulation tool is NS2, with testing carried out three scenarios covering QoS analysis of throughput, packet loss, delay jitter querying. Then the results of this analysis can be used as a reference in the design and management of VoIP services in accordance with existing needs.

Item Type: Thesis (Other)
Subjects: T Technology > TA Engineering (General). Civil engineering (General)
Divisions: Faculty of Engineering > Department of Informatics (55201)
Depositing User: Halimatus Zahroh
Date Deposited: 17 Feb 2015 11:56
Last Modified: 17 Feb 2015 11:56
URI : http://eprints.umm.ac.id/id/eprint/16325

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