Please use this identifier to cite or link to this item: http://hdl.handle.net/10603/215958
Title: Performance Enhancement Techniques for Voice Over IP Telephony
Researcher: Saira Banu A
Guide(s): Dr. R. SHRIRAM
Keywords: Performance Enhancement, Voice, IP Telephony
University: B S Abdur Rahman University
Completed Date: 20/08/2018
Abstract: ABSTRACT newlineVoice over Internet Protocol (VoIP) is becoming popular in the recent years because of its advantage of low-priced calls when compared with the existing Public Switch Telephone Network. The main reason for the invention of VoIP is to make long distance telephone calls at low cost. But this VoIP service is misused by the advertisers, prank callers, telemarketers and call centers by generating bulk voice and text traffic. This bulk traffic will degrade the performance of other users in the network because of queuing delay and congestion. The performance of voice service is degraded because unreceived and destroyed packets are not sent again and considered as lost packets. So maintaining the voice service quality and voice spam detection has become the major challenge in the mobile VoIP service. newlineThe primary objective of this research work focuses on enhancing the quality of voice over the VoIP in three approaches based on their functionality: newline[1] Modulation based approach that that uses the modulation selection algorithm for transmitting the data. newline[2] Network based approach that uses the jitter buffer management algorithm and packet loss concealment algorithm to handles the problems because of dropped and delayed packets in the network. newline[3] Spammer based approach that uses the spam detection algorithm to detect the voice spammers. newlineThese approaches are addressed using the following proposed algorithms: newline1. Best effort Modulation and Coding Scheme (BE-MCS) newline2. Best Effort Jitter buffer management Algorithm (BE-JBMA) newline3. Effective Packet Loss Concealment Algorithm(E-PLCA) newline4. Spam caller detection algorithm newlineThe proposed research work increases the throughput, reduces the packet drop ratio in the jitter buffer, improves the quality of reconstructed speech signal and identifies the Spam callers in the network. The integration of the proposed algorithms enhances the performance of the real-time voice over the IP telephony. newline
Pagination: 126
URI: http://hdl.handle.net/10603/215958
Appears in Departments:Department of Computer Science and Engineering

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