Abstract—VoIP and internet users are more and more interested in the multimedia communication services and this has made the academic researches in this domain to be risen dramatically. Media Servers play a key role in multiple communications and voice and video conferencing. Thus, implementation and evaluation of media servers has a great importance in the world of VoIP and Internet Telephony. In this paper we configured an open source media server called SEMS to run a voice conference. Then we evaluated our conference server using CPU load, packet loss and jitter. We carried out our tests using different speech Codecs like G.711, G.726, GSM, and iLBC and different number of participants. We also use an open source traffic generator called SIPp in order to generate SIP and RTP traffic simulating the participants of the conference. We show when the number of participants in conference exceeds a certain value that is called capacity of server then the quality of communication decreases dramatically where an overload control is needed. The capacity of the conference media server is evaluated and obtained in this report for several Codecs and deployment situations.
Index Terms—SEMS, SIP, CPU load, codecs.
Fatemeh Samsami was with Iran University of Science and Technology, Tehran, Iran. The major field of study is Information and Communications Technology (ICT) (e-mail: samsami@vu.iust.ac.ir).
Ahmad Akbari is with the Faculty of Computer Engineering in Iran University of Science and Technology, Tehran, Iran (e-mail: akbari@iust.ac.ir).
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Cite:Fatemeh Samsami and Ahmad Akbari, "Evaluation of Media Server in SIP-Based Voice Conferencing," International Journal of Computer Theory and Engineering vol. 5, no. 4, pp. 745-749, 2013.
