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Advanced implementation of IP telephony at Czech universities
- WSEAS Transactions on Communications, Volume 9, Issue
, 2010
"... Abstract:- IP telephony is convenient way of communication and brings number of benefits. One of them is the fact that many services can be implemented in a original way and it opens new field of research and is a challenge for designers of communication solutions especially based on open-source pla ..."
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Abstract:- IP telephony is convenient way of communication and brings number of benefits. One of them is the fact that many services can be implemented in a original way and it opens new field of research and is a challenge for designers of communication solutions especially based on open-source platform. This article describes proposal and advanced implementation of IP telephony which have been arisen in Czech academical environment, concretely in a group of IP telephony acting under Czech Education and Scientific Network association. Author is a senior researcher in this group and he decided to describe the most considerable voice over IP implementations at Czech universities and share knowledge with other experts interesting in IP telephony.
Simulation-based Optimization of Signaling Procedures in IP Multimedia Subsystem
"... Abstract—This paper presents a simulation-based optimization of signaling procedures in Internet protocol Multimedia Subsystem (IMS). The aim is to improve the performance of signaling procedures by applying an algorithm for Session Initiation Protocol (SIP) message classification and prioritization ..."
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Abstract—This paper presents a simulation-based optimization of signaling procedures in Internet protocol Multimedia Subsystem (IMS). The aim is to improve the performance of signaling procedures by applying an algorithm for Session Initiation Protocol (SIP) message classification and prioritization that is proposed in our previous work. This three-priority level classification of SIP messages is implemented in Network Simulator version 2 (ns-2). Its effectiveness is verified through the simulation-based study of SIP signaling procedures under different conditions. The simulation results are analyzed in terms of Registration
INTERNATIONAL JOURNAL OF MATHEMATICAL MODELS AND METHODS IN APPLIED SCIENCES Performance Analysis of Virtualized Real-time Applications
"... Abstract — This article deals with the impact of virtualization techniques on interactive delay-sensitive applications running in realtime, particularly IP telephony. Many institutions, organizations and home users often adopt the virtualized solutions for their safety, ease of administration and ba ..."
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Abstract — This article deals with the impact of virtualization techniques on interactive delay-sensitive applications running in realtime, particularly IP telephony. Many institutions, organizations and home users often adopt the virtualized solutions for their safety, ease of administration and backup. Virtualization, which was chiefly the prerogative of companies and the academic world in its early days, has gradually develop its platform to reach out to the ordinary users who can benefit from running virtual machines. The aim of this paper is to examine the impact of a virtual machine on real-time traffic, in our case IP telephony based on the SIP and the RTP, which are now the cornerstone of VoIP technology. This article also analyses the impact of memory size and the number of processor cores on the delay itself and its variance, thus allowing user to have full picture when deciding what virtualization tool to use and how to configure so it performs the best possible way.
Original language: EN Original title:
"... Produced by CESNET led working group on Multimedia transmissions and collaborative environment (CBPD142) ..."
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Produced by CESNET led working group on Multimedia transmissions and collaborative environment (CBPD142)
SIP-Based QoS in IP Telephony
"... Abstract—In this paper, the authors analyze the factors resulting to the degradation of the quality of service in voice over IP (VoIP) telephony. SIP protocol is used to explore the QoS characteristics of different Codec. The system is designed for an IP telephony service provider with other GSM ope ..."
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Abstract—In this paper, the authors analyze the factors resulting to the degradation of the quality of service in voice over IP (VoIP) telephony. SIP protocol is used to explore the QoS characteristics of different Codec. The system is designed for an IP telephony service provider with other GSM operator network cloud. A soft IP PBX maintains the dial pattern, SIP proxies are used in order to implement call control and to allow the distributing of the gateways ’ lines. The system is executed in a test bed where QoS factors likes delay, packet loss, forward and reverse jitter, MOS and delta measured. A Popular simulation software Wireshark is used to simulate and analysis the transmission characteristics of packetized voice over the IP network channel. The results are graphically shown, which reveal the effects of packet size on QoS. Packet sizes from 10 to 60 bytes are used for a G.729 Codec in 11.2-24 Kbps channel bandwidths are studied. A 20 byte packet size at 24 Kbps channel bandwidth is seen to show the best result in terms of jitter, delay and delta. The results of studies conducted by also shown that G.729 codec shown high MOS value and low mean forward and reverse jitter for simultaneous call compare with other codecs.
