Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server

Sajad Gharaguozloo; Abdolhamid Zahedi; Mohammad Norouzi; Hamid Chegini

doi:doi:10.11648/j.wcmc.20160402.11

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Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server

Sajad Gharaguozloo, Abdolhamid Zahedi, Mohammad Norouzi, Hamid Chegini

Published in International Journal of Wireless Communications and Mobile Computing (Volume 4, Issue 2)

Received: 21 February 2016 Accepted: 1 March 2016 Published: 12 March 2016

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Abstract

By development of multi-media in networks, the borders among networks are changed and all networks are approaching to be united. Unity of data, video and voice networks in one network has many advantages and disadvantages for users and servers. One of the disadvantages is unwanted events in network including load increase, jitter, information packet loss, delay and etc. and all these lead into low quality of voice and disconnection during simultaneous call. The use of multimedia server is one of the efficient ways to improve VOIP. We can enable the video conferences to transit information packets by media servers, so we can say: media servers can be used as core component for VOIP. In this research work, assessing of media servers is done by simulators that they produce RTP's connections, in additions as an experimental components SEMS that it's a source of media server, is used for asserting the quality by doing packets information with SIP. We can observe that the more increase of connections and pass of the certain threshold, the less of quality. In addition, the other performance metrics such as error rate And packet lost are asserted. The identification of load saturation points and the efforts to eliminate disturbing factors during the increase of simultaneous call in this telephone system, can present quality-based approach to servers of these networks.

Published in	International Journal of Wireless Communications and Mobile Computing (Volume 4, Issue 2)
DOI	10.11648/j.wcmc.20160402.11
Page(s)	12-17
Creative Commons	This is an Open Access article, distributed under the terms of the Creative Commons Attribution 4.0 International License (http://creativecommons.org/licenses/by/4.0/), which permits unrestricted use, distribution and reproduction in any medium or format, provided the original work is properly cited.
Copyright	Copyright © The Author(s), 2024. Published by Science Publishing Group

Keywords

VOIP, SEMS, SIP, Increase of Simultaneous Call

References

[1]	H. Wook, S. Kang, D. Kim, “Performance Enhancement of SIP proxy server by using Ihash for matching transaction”, IEEE, ISBN 978-89-5519-131-8 93560, Feb 2007
[2]	Mauro Femminella, Roberto Francescangeli, Francesco Giacinti, Emanuele Maccherani, “Design, Implementation, and performance of an advanced SIP-based call control for VoIP services”, IEEE, ISBN 978-1-4244-3435, 2009
[3]	http://sipp.sourceforge.net/doc/reference.html
[4]	Montoro, P, Casilari, E, 2009, A Comparative Study of VoIP Standards with Asterisk, Fourth International Conference on Digital Telecommunications
[5]	Iseki, F, Sato, Y, Kim, M. 2011, VoIP System based on Asterisk forEnterprise Network, ICACT2011
[6]	Pantelis A. Frangoudisa, George C. Polyzosb, On the performance of secure user-centric VoIP communication, 2014, Computer Networks Volume 70, 9 September Pages 330–344
[7]	Abdul Qadeer, M, Shah, K, Goel, U, 2012, Voice - Video Communication on Mobile Phonesand PCs’ using Asterisk EPBX, International Conference on Communication Systems and Network Technologies
[8]	WWW.VOIP-IRAN.COM
[9]	Nikos Vrakasa, Dimitris Geneiatakisb, Costas Lambrinoudakisa, Obscuring users' identity in VoIP/IMS environme, 2014, Computers & Security, Volume 43, June, Pages 145–158
[10]	Ryan Farley, Xinyuan Wang, Exploiting VoIP softphone vulnerabilities to disable host computers: Attacks and mitigation, 2014, International Journal of Critical Infrastructure Protection, Volume 7, Issue 3, September, Pages 141–154
[11]	Liping Zhang, Shanyu Tang, Shaohui Zhu, An energy efficient authenticated key agreement protocol for SIP-based green VoIP networks, 2016, Journal of Network and Computer Applications, Volume 59, January, Pages 126–133
[12]	Jinzhu Wanga, et al, Probe-based end-to-end overload control for networks of SIP servers, 2014, Journal of Network and Computer Applications, Volume 41, May, Pages 114–125
[13]	Basicevic, M. Popovic, D. Kukolj, 2008, Comparison of SIP and H. 323 Protocols, Proc. of The Third International Conference on Digital Telecommunications (ICDT’08), Bucharest (Romania), Jul. pp.162-167
[14]	Regis J. (Bud) Bates, Chapter 6 – Other protocols SRTP, ZRTP, and SIPS, 2015, Securing VOIP Keeping your VOIP Network Safe, Pages 123–150
[15]	J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002
[16]	J. Rosenberg, "A Framework for Conferencing with the Session Initiation Protocol (SIP)", RFC 4353, February 2006
[17]	R. Even, N. Ismail, "Conferencing Scenarios", RFC 4597, August 2006
[18]	http://www.iptel.org/sems.
[19]	C. Partridge, “Isochronous Applications Do Not Require Jitter-Controlled Networks”, RFC 1257, September 1991
[20]	H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", RFC 3550, July 2003
[21]	A. H. Ashouri, F. Samsami, A. Akbari, “E-Learning Media Server Evaluation and its architecture modeling with signaling load tests,” ICeLT, IUST, Tehran, Iran, Dec 2009
[22]	A. H. Ashouri. “Media Server Evaluation and Real-Time Tests” Iran University of Science and Technology, B. Sc Thesis, p46-61, Sep 2009

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APA Style

Sajad Gharaguozloo, Abdolhamid Zahedi, Mohammad Norouzi, Hamid Chegini. (2016). Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server. International Journal of Wireless Communications and Mobile Computing, 4(2), 12-17. https://doi.org/10.11648/j.wcmc.20160402.11

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ACS Style

Sajad Gharaguozloo; Abdolhamid Zahedi; Mohammad Norouzi; Hamid Chegini. Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server. Int. J. Wirel. Commun. Mobile Comput. 2016, 4(2), 12-17. doi: 10.11648/j.wcmc.20160402.11

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AMA Style

Sajad Gharaguozloo, Abdolhamid Zahedi, Mohammad Norouzi, Hamid Chegini. Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server. Int J Wirel Commun Mobile Comput. 2016;4(2):12-17. doi: 10.11648/j.wcmc.20160402.11

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@article{10.11648/j.wcmc.20160402.11,
  author = {Sajad Gharaguozloo and Abdolhamid Zahedi and Mohammad Norouzi and Hamid Chegini},
  title = {Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server},
  journal = {International Journal of Wireless Communications and Mobile Computing},
  volume = {4},
  number = {2},
  pages = {12-17},
  doi = {10.11648/j.wcmc.20160402.11},
  url = {https://doi.org/10.11648/j.wcmc.20160402.11},
  eprint = {https://article.sciencepublishinggroup.com/pdf/10.11648.j.wcmc.20160402.11},
  abstract = {By development of multi-media in networks, the borders among networks are changed and all networks are approaching to be united. Unity of data, video and voice networks in one network has many advantages and disadvantages for users and servers. One of the disadvantages is unwanted events in network including load increase, jitter, information packet loss, delay and etc. and all these lead into low quality of voice and disconnection during simultaneous call. The use of multimedia server is one of the efficient ways to improve VOIP. We can enable the video conferences to transit information packets by media servers, so we can say: media servers can be used as core component for VOIP. In this research work, assessing of media servers is done by simulators that they produce RTP's connections, in additions as an experimental components SEMS that it's a source of media server, is used for asserting the quality by doing packets information with SIP. We can observe that the more increase of connections and pass of the certain threshold, the less of quality. In addition, the other performance metrics such as error rate And packet lost are asserted. The identification of load saturation points and the efforts to eliminate disturbing factors during the increase of simultaneous call in this telephone system, can present quality-based approach to servers of these networks.},
 year = {2016}
}

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TY  - JOUR
T1  - Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server
AU  - Sajad Gharaguozloo
AU  - Abdolhamid Zahedi
AU  - Mohammad Norouzi
AU  - Hamid Chegini
Y1  - 2016/03/12
PY  - 2016
N1  - https://doi.org/10.11648/j.wcmc.20160402.11
DO  - 10.11648/j.wcmc.20160402.11
T2  - International Journal of Wireless Communications and Mobile Computing
JF  - International Journal of Wireless Communications and Mobile Computing
JO  - International Journal of Wireless Communications and Mobile Computing
SP  - 12
EP  - 17
PB  - Science Publishing Group
SN  - 2330-1015
UR  - https://doi.org/10.11648/j.wcmc.20160402.11
AB  - By development of multi-media in networks, the borders among networks are changed and all networks are approaching to be united. Unity of data, video and voice networks in one network has many advantages and disadvantages for users and servers. One of the disadvantages is unwanted events in network including load increase, jitter, information packet loss, delay and etc. and all these lead into low quality of voice and disconnection during simultaneous call. The use of multimedia server is one of the efficient ways to improve VOIP. We can enable the video conferences to transit information packets by media servers, so we can say: media servers can be used as core component for VOIP. In this research work, assessing of media servers is done by simulators that they produce RTP's connections, in additions as an experimental components SEMS that it's a source of media server, is used for asserting the quality by doing packets information with SIP. We can observe that the more increase of connections and pass of the certain threshold, the less of quality. In addition, the other performance metrics such as error rate And packet lost are asserted. The identification of load saturation points and the efforts to eliminate disturbing factors during the increase of simultaneous call in this telephone system, can present quality-based approach to servers of these networks.
VL  - 4
IS  - 2
ER  -

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Author Information

Sajad Gharaguozloo

Telecommunication, of Non-profit Institution of Higher Education, ABA, Abyek, Qazvin, Iran
Abdolhamid Zahedi

Non-profit Institution of Higher Education, ABA, Abyek, Qazvin, Iran
Mohammad Norouzi

Non-profit Institution of Higher Education, ABA, Abyek, Qazvin, Iran
Hamid Chegini

Non-profit Institution of Higher Education, ABA, Abyek, Qazvin, Iran

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Plain Text BibTeX RIS

APA Style

Sajad Gharaguozloo, Abdolhamid Zahedi, Mohammad Norouzi, Hamid Chegini. (2016). Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server. International Journal of Wireless Communications and Mobile Computing, 4(2), 12-17. https://doi.org/10.11648/j.wcmc.20160402.11

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ACS Style

Sajad Gharaguozloo; Abdolhamid Zahedi; Mohammad Norouzi; Hamid Chegini. Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server. Int. J. Wirel. Commun. Mobile Comput. 2016, 4(2), 12-17. doi: 10.11648/j.wcmc.20160402.11

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AMA Style

Sajad Gharaguozloo, Abdolhamid Zahedi, Mohammad Norouzi, Hamid Chegini. Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server. Int J Wirel Commun Mobile Comput. 2016;4(2):12-17. doi: 10.11648/j.wcmc.20160402.11

Copy | Download

@article{10.11648/j.wcmc.20160402.11,
  author = {Sajad Gharaguozloo and Abdolhamid Zahedi and Mohammad Norouzi and Hamid Chegini},
  title = {Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server},
  journal = {International Journal of Wireless Communications and Mobile Computing},
  volume = {4},
  number = {2},
  pages = {12-17},
  doi = {10.11648/j.wcmc.20160402.11},
  url = {https://doi.org/10.11648/j.wcmc.20160402.11},
  eprint = {https://article.sciencepublishinggroup.com/pdf/10.11648.j.wcmc.20160402.11},
  abstract = {By development of multi-media in networks, the borders among networks are changed and all networks are approaching to be united. Unity of data, video and voice networks in one network has many advantages and disadvantages for users and servers. One of the disadvantages is unwanted events in network including load increase, jitter, information packet loss, delay and etc. and all these lead into low quality of voice and disconnection during simultaneous call. The use of multimedia server is one of the efficient ways to improve VOIP. We can enable the video conferences to transit information packets by media servers, so we can say: media servers can be used as core component for VOIP. In this research work, assessing of media servers is done by simulators that they produce RTP's connections, in additions as an experimental components SEMS that it's a source of media server, is used for asserting the quality by doing packets information with SIP. We can observe that the more increase of connections and pass of the certain threshold, the less of quality. In addition, the other performance metrics such as error rate And packet lost are asserted. The identification of load saturation points and the efforts to eliminate disturbing factors during the increase of simultaneous call in this telephone system, can present quality-based approach to servers of these networks.},
 year = {2016}
}

Copy | Download

TY  - JOUR
T1  - Presenting Solutions to Increase Simultaneous Call in VOIP System by SIP Protocol - Based Media Server
AU  - Sajad Gharaguozloo
AU  - Abdolhamid Zahedi
AU  - Mohammad Norouzi
AU  - Hamid Chegini
Y1  - 2016/03/12
PY  - 2016
N1  - https://doi.org/10.11648/j.wcmc.20160402.11
DO  - 10.11648/j.wcmc.20160402.11
T2  - International Journal of Wireless Communications and Mobile Computing
JF  - International Journal of Wireless Communications and Mobile Computing
JO  - International Journal of Wireless Communications and Mobile Computing
SP  - 12
EP  - 17
PB  - Science Publishing Group
SN  - 2330-1015
UR  - https://doi.org/10.11648/j.wcmc.20160402.11
AB  - By development of multi-media in networks, the borders among networks are changed and all networks are approaching to be united. Unity of data, video and voice networks in one network has many advantages and disadvantages for users and servers. One of the disadvantages is unwanted events in network including load increase, jitter, information packet loss, delay and etc. and all these lead into low quality of voice and disconnection during simultaneous call. The use of multimedia server is one of the efficient ways to improve VOIP. We can enable the video conferences to transit information packets by media servers, so we can say: media servers can be used as core component for VOIP. In this research work, assessing of media servers is done by simulators that they produce RTP's connections, in additions as an experimental components SEMS that it's a source of media server, is used for asserting the quality by doing packets information with SIP. We can observe that the more increase of connections and pass of the certain threshold, the less of quality. In addition, the other performance metrics such as error rate And packet lost are asserted. The identification of load saturation points and the efforts to eliminate disturbing factors during the increase of simultaneous call in this telephone system, can present quality-based approach to servers of these networks.
VL  - 4
IS  - 2
ER  -

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