Abstract
Voice-over Internet Protocol (VoIP) telephony is increasingly becoming a variant telecommunication technology that one day may surpass the old analog and digital telephone systems. The Quality-of-Service (QoS) factor is an important parameter to be considered when measuring the performance of a VoIP system. Many factors may influence the QoS of a given VoIP system. Some of these factors are delay, jitter and packet loss. This paper served to demonstrate that the algorithmic delay (latency) of the Pulse Code Modulation (PCM) speech compression algorithm which has originally been implemented in software can be significantly reduced by implementation in hardware. The PCM speech compression algorithm is first implemented, verified to demonstrate equivalence and then validated by comparing the latency of the hardware-implemented speech compression algorithm with that of an existing software implementation.
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