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Session initiation protocol improvement using inter-asterisk exchange

Published: 05 December 2011 Publication History

Abstract

Within the VolP networks environment, there are three protocols that solve the problem of voice packet signaling, known as "highlight protocols": H323, SIP and IAX. Particularly, this document focuses on a specific difference between SIP and IAX: the bandwidth, which is an essential parameter in order to design and optimize a VoIP network for an organization. One of the most important factors to consider when building VoIP networks is the proper capacity planning. About capacity planning, bandwidth calculation is an important factor to consider when designing and troubleshooting packet voice networks for good voice quality. This paper presents important explanations about the bandwidth utilization over VoIP networks; this include o take some theoretical foundations and calculate the bandwidth per media flow for a VoIP trunk, in which different voice codecs is enabled (specifically G. 711 and JSM) and when a VoIP protocol such as SIP and IAX. Voice over IP (VoIP) is used. These calculations will be compared here against the results obtained stats about traffic analysis from some network analyzers over a physical Ethernet trunk between two Asterisk servers into a specific LAN, simulating two different branches for an organization. Both the analysis of the results obtained and the proper conclusions from this work, are useful when calculating he maxim number of simultaneous calls or the minimal capability of a data link, necessary for a single number of voice conversations, taking into account the audio codec's used and signaling and data flow protocols.

References

[1]
M. Spencer, B. Capouch, E. Guy, "RFC 5456 for IAX", http://www.rfc-editor.org/authors/rfc5456.txt
[2]
RFC 3261 - SIP: Session Initiation Protocol. J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler. June 2002
[3]
T. Abbasi, S. Prasad, N. Seddigh, I. Lambadaris, "A comparative study of the sip and iax voip protocols", 2007.
[4]
K. Chava, J. Ilow; "Integration of Open Source and Enterprise IP PBXs"; Dalhousie University, Halifax, NS, Canada.
[5]
P. Montoro, E. Casilari; "A Comparative Study of VoIP Standards with Asterisk"; Dpto. Tecnologia Electrónica, University of Malaga (Spain); 2009 Fourth International Conference on Digital Telecommunications
[6]
Protocol Analyzer for IAX2; http://www.unleashnetworks.com/resources/articles/72-asterisk-call-analyzer-for-iax2.html
[7]
Asterisk Auto-dial; http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
[8]
Análisis de paquetes IAX en la herramienta Wireshark; http://wiki.wireshark.org/IAX2
[9]
Cisco Systems; Bandwidth per call; http://www.cisco.com/en/US/tech/tk652/tk698/technologiestechnote09186a0080094ae2.shtml#topic1
[10]
F. E. Goncalves, "Asterisk PBX guía de la configuración", Tercera generación, 1a edición, Rio de Janeiro 2007
[11]
Voice/Data, Cisco Systems; Article of Cisco Systems.
[12]
J. T. Da Loligo; Cálculo de ancho de banda para IAX modo trunked; http://www.voip-info.org/wiki/wiki-Asterisk+bandwidth+iax2
[13]
Vyatta Open Networking: Software, productos y manuales de instalación y configuración; http://www.vyatta.com/
[14]
Instalación CentOS 5.1 para centrales telefónicas Open Source Asterisk PBX; http://192.168.222.5/wiki/index.php/InstalacionsCentOS 5.1 para Asterisk - GNU
[15]
Instalación y configuración de Asterisk PBX en Ubuntu Server; http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Paso
[16]
Instalación y configuración de Asterisk sobre CENTOS 5.2; http://wiki.millenium.com.co/wiki/index.php/Instalacion Asterisk 1.4.X
[17]
R. Gallego, Transmisión de Audio, QoS en redes de nueva generación, Departamento de Ingeniería Electrónica y Comunicaciones, Universidad de Zaragoza, 2005.
[18]
Informatik - Rechnerbetrieb; "US Robotics GSM"; http://user.cs.tu-berlin.de/~jutta/gsm/private/usr-gsm.html; Alemania, 2009.
[19]
G. Gorrotxategui, I. Baz, "Curso sobre VoIP y Asterisk Módulo 1", IRONTEC, 2002, http://www.irontec.Com

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        cover image ACM Other conferences
        iiWAS '11: Proceedings of the 13th International Conference on Information Integration and Web-based Applications and Services
        December 2011
        572 pages
        ISBN:9781450307840
        DOI:10.1145/2095536
        Permission to make digital or hard copies of all or part of this work for personal or classroom use is granted without fee provided that copies are not made or distributed for profit or commercial advantage and that copies bear this notice and the full citation on the first page. Copyrights for components of this work owned by others than ACM must be honored. Abstracting with credit is permitted. To copy otherwise, or republish, to post on servers or to redistribute to lists, requires prior specific permission and/or a fee. Request permissions from [email protected]

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        New York, NY, United States

        Publication History

        Published: 05 December 2011

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        Author Tags

        1. IAX
        2. SIP
        3. VoIP
        4. audio codec's
        5. bandwidth

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