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The quality of speech signal over a VoIP system is degraded by various network layer problems which include jamming, jitter, packet loss. Different types of noises also degrade the quality of speech signal such as external noise and quantization noise. This paper improves the quality of VoIP speech signal affected by these noises and network layer problems. The quality of degraded VoIP speech signal coded by using ITU-T G.711 audio coding standard and implemented on the TMS320C6713 DSK has been compared with the quality of VoIP speech signal coded by using G.729 audio data compression standard which uses code-excited linear prediction speech coding (CS-ACELP) for coding purpose and is implemented on TMS320C6713 DSK.
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