A Research of VoIP Jitter on Packet Loss in GSM Voice over IP Systems
Kehinde Adebusuyi1, Ezea Hilary2, Gerald Ijemaru3
1Kehinde Adebusuyi, Department of Electrical and Electronics Engineering, Federal University Oye-Ekiti, Ekiti State, Nigeria.
2Ezea Hilary, Department of Electrical and Electronics Engineering, Federal University Oye-Ekiti, Ekiti State, Nigeria.
3Gerald Ijemaru, Department of Electrical and Electronics Engineering, Federal University Oye-Ekiti, Ekiti State, Nigeria.
Manuscript received on 14 August 2017 | Revised Manuscript received on 20 August 2017 | Manuscript Published on 30 August 2017 | PP: 5-12 | Volume-6 Issue-12, August 2017 | Retrieval Number: L24520861217/17©BEIESP
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© The Authors. Blue Eyes Intelligence Engineering and Sciences Publication (BEIESP). This is an open access article under the CC-BY-NC-ND license (http://creativecommons.org/licenses/by-nc-nd/4.0/)
Abstract: Jitter in the next generation network Voice over Internet Protocol Systems is a fundamental network problem in call quality measurement research. We classify Jitter according to call quality requirement in overall quality of service of a reliable information and communication technology infrastructure especially as convergence of voice, video and data increases in Internet telephony systems. Internet telephony mode of communication systems is delivered via Voice over IP media and signaling techniques adopted for internet transmission in achieving VOIP system. Our goal in this paper is to investigate the impact of VoIP Jitter by studying the optimal packet Call Flow Routing (CFR) model in Real Voice Optimization to reduce Jitter in VoIP systems. The results of the simulations shows that as the time in arrival of consecutive voice packets increase, the optimal packet Call Flow Programmatic Routing (CFR) reduces with time varying demands in the total packetized call transmission across all network links. We observed that the set of call Flow Routing constraints investigated imposes the packet flow load balance at end-to-end gateway router and shows a reduction in packet loss and at a point, records no loss. The value at which the time slots for the network caller party demand increases, more Caller party serves the subsets of packet flows by simulating the behavior of the network per each slot independently of each slots. Hence, the caller router and called party router flows are positive by resolving the individual routing problem per each slot and integrating the models.
Keywords: Jitter, VoIP, Voice, Optimize CFR, Packet Flow
Scope of the Article: Expert Systems