Performance analysis of speech codec (GSM, ILBC, SPEEX) for voip over wireless local area network (WLAN) with respective signal to noise-ratio (SNR)
Voice over Internet Protocol (VoIP) is one of the fastest growing Internet applications. It is a viable alternative to the traditional telephony systems due to its high resource utilization and cost efficiency. Meanwhile, Wireless Local Area Networks (WLANs) have become a ubiquitous networking techn...
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Main Author: | |
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Format: | Undergraduates Project Papers |
Language: | English |
Published: |
2013
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Subjects: | |
Online Access: | http://umpir.ump.edu.my/id/eprint/8744/1/cd8314.pdf http://umpir.ump.edu.my/id/eprint/8744/ |
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Summary: | Voice over Internet Protocol (VoIP) is one of the fastest growing Internet applications. It is a viable alternative to the traditional telephony systems due to its high resource utilization and cost efficiency. Meanwhile, Wireless Local Area Networks (WLANs) have become a ubiquitous networking technology that has been deployed around the world. In this research, 3 types of speech codec (GSM, ILBC, SPEEX) in the same sampling rate of (11-13) kbps are chosen to be test in predefined network environments to measure the performance base on R-Factor, MOS, and packet jitter and packet loss. Thus, a codec is expected to provide good quality of VoIP. And in some circumstances, bandwidth may be a crucial factor between the success and failure of an application. With the likes of Internet applications such as video and audio streaming, video and audio downloading, these has contributed to the increase of Internet users and which directly affect the performance of speech codec when tested with other traffic in the network because it were using the same network bandwidth. All three mention speech codec will be test based on these criteria. The speech quality of three speech codec namely GSM (13kbps) , ILBC (13.33 kbps) , and Speex (11kbps) under various network performance based on pre-determined SNR values will be evaluated and compare against. Several tests are constructed to prove that it meets the interest of investigation. The experimental procedure of this dissertation can be summarized to 2 main experiments which need to be repeated for each speech codec and for each predefined SNR value. Both types of network on two way communication testing; 1) Optimum Network, and 2) Network with others traffic, need to be repeated for all three speech codec GSM, ILBC, Speex with each respective SNR values; 10 dB, 20 dB, and 30 dB. All test criteria will be carry out on real devices simulation. At the end, the performance measurement of VOIP on Quality of Services; such as MOS, R-Factor, packet loss and packet jitter will be observe to determine the best speech codec in each scenario. |
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