Study of enhanced VoIP performance under congested wireless network scenarios

Mukhopadhyay, Atri; Chakraborty, Tamal; Bhunia, Suman; Misra, Iti Saha; Sanyal, Sugata

doi:10.1109/comsnets.2011.5716509

Cited by 10 publications

(7 citation statements)

References 8 publications

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“…Alternatively, UDP does not apply ACK scheme, thus it is more suitable for VoIP applications. However, since UDP lacks congestion control techniques, RTP and RTCP protocols were designed to cover these problems [2].…”

Section: Voip Protocolsmentioning

confidence: 99%

“…R > 100 1 + 0.035 R + 7.10 -6 R(R-60)×(100-R) , 0 ≤ R ≤ 100 (2) R-factor can be also compared to their corresponding values of MOS and user satisfaction category as shown in TABLE II [19]. OPNET (Optimized Network Engineering Tool) Modeler is a powerful and discrete event based network modeling and simulation tool with a wide variety of features and a rich library of models, protocols, graphical results, etc.…”

Section: Performance Evaluation Technique: E-modelmentioning

confidence: 99%

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Studying the effect of transmission rate and packet size parameters on VoIP performance

Alshakhsi

Hasbullah

2012

2012 International Conference on Computer &Amp; Information Science (ICCIS)

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In recent years, VoIP over Wireless Local Area Networks (VoWLAN) technology has been increasingly on demand due to its advantages such as it is cheaper and supports mobility. Yet, the required stringent service quality for VoIP applications particularly in WLAN is still a challenge. This paper investigates the effect of two key parameters on VoIP quality; transmission/data rate and packet size. In this study, the data rate value can be 11, 5.5, 2, or 1 Mbps according to IEEE802.11b WLAN, while packet size is determined by the number of frames inserted in the packet. A simulation study has been implemented using OPNET Modeler. The first part of the results compares VoIP quality values obtained for each data rate. It also shows how decreasing the data rate degrades the VoIP performance. On the other hand, the second part of the results demonstrates how changing voice packet size can significantly affect VoIP quality. Further, it is discussed and analyzed the effect of packet size on delay, packet loss and network bandwidth requirement which accordingly affects VoIP quality.

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Section: Voip Protocolsmentioning

confidence: 99%

Section: Performance Evaluation Technique: E-modelmentioning

confidence: 99%

Studying the effect of transmission rate and packet size parameters on VoIP performance

Alshakhsi

Hasbullah

2012

2012 International Conference on Computer &Amp; Information Science (ICCIS)

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“…For example in our implementation, as wireless calls are being made, parameter optimizations of APs as described in [40] can also be implemented. Moreover, techniques for enhancing VoIP performance under congested scenarios can also be incorporated in this work, for example, by varying the packet payload size as suggested in [41]. Advanced artificial intelligence search techniques can be further applied.…”

Section: Benefitsmentioning

confidence: 99%

VOIP Call Optimization in Diverse Network Scenarios Using Learning Based State-Space Search Technique

Chakraborty¹,

Mukhopadhyay²,

Misra³

et al. 2011

IJWMN

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A VoIP based call has stringent QoS requirements with respect to delay, jitter, loss, MOS and R-Factor. Various QoS mechanisms implemented to satisfy these requirements must be adaptive under diverse network scenarios and applied in proper sequence, otherwise they may conflict with each other. The objective of this paper is to address the problem of adaptive QoS maintenance and sequential execution of available QoS implementation mechanisms with respect to VoIP under varying network conditions. In this paper, we generalize this problem as state-space problem and solve it. Firstly, we map the problem of QoS optimization into state-space domain and apply incremental heuristic search. We implement the proposed algorithm under various network and user scenarios in a VoIP test-bed for QoS enhancement.Then learning strategy is implemented for refinement of knowledge base to improve the performance of call quality over time. Finally, we discuss the advantages and uniqueness of our approach.

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“…The lost packets usually are not retransmitted due to real-time service requirements of VoIP. Hence, the transmission of voice stream goes through a serious packet loss in VoIP, causing gaps in the IP media stream [1] [2]. Group packet loss of information not only directly reduces the voice quality, but also causes the spread of mistaken frame because of the correlation between the adjacent frames in the phoneticsource encoding.…”

Section: Introductionmentioning

confidence: 99%

Packet-level Reed-Solomon codes over burst-loss channels

Che

Bai

Feng

2012

2012 IEEE 14th International Conference on Communication Technology

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The packet-level Reed-Solomon (PRS) code can help to recover burst loss packets for VoIP service over wireless networks. A detailed study on PRS code over a burst loss channel is carried out by means of simulation in this paper.The Gilbert model based on a two-state Markov chain is established for the burst-loss channel. The encoding and decoding algorithms of the PRScode including the truncated code are given. The performance of different PRS codes with three burst-loss channel parameters is obtainedand the relationship between the improved performance and the PRS codes over burst-loss channels is discussed. Based on our study, a method of choosing the suitable PRS codes over burst-loss channel is advised at the end of the paper.

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Study of enhanced VoIP performance under congested wireless network scenarios

Cited by 10 publications

References 8 publications

Studying the effect of transmission rate and packet size parameters on VoIP performance

Studying the effect of transmission rate and packet size parameters on VoIP performance

VOIP Call Optimization in Diverse Network Scenarios Using Learning Based State-Space Search Technique

Packet-level Reed-Solomon codes over burst-loss channels

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