A novel method of compressing speech with higher bandwidth efficiency This uses a novel method of speech compression and transmission. This method saves the transmission bandwidth required for the speech signal by a considerable amount. This scheme exploits the property of low pass nature of the speech signal. Also this method applies equally well for any signal, which is low pass in nature, speech being the more widely used in Real Time Communication, is highlighted here. As per this method, the low pass signal (speech) at the transmitter is divided into set of packets, each containing, say N number of samples. Of the N samples per packet, only certain lesser number of samples, say N alone are transmitted. Here is less than unity, so compression is achieved. The N samples per packet are subjected to a N-Point DFT. Since low pass signals alone are considered here, the number of significant values in the set of DFT samples is very limited. Transmitting these significant samples alone would suffice for reliable transmission. The number of samples, which are transmitted, is determined by the parameter . The parameter is almost independent of the source of the speech signal. In other methods of speech compression, the specific characteristics of the source such as pitch are important for the algorithm to work. An exact reverse process at the receiver reconstructs the samples. At the receiver, the N-point IDFT of the received signal is performed after necessary zero padding. Zero padding is necessary because at the transmitter of the N samples only N samples are transmitted, but at the receiver N samples are again needed to honestly reconstruct the signal