The voice pass band for analog telephony voice signals is defined as 300 through 3300 Hertz. Any signal carried on a telephone circuit within that range is defined as an "in-band" signal. This characterizes all speech signals. Digital voice is arrived at by sampling the 4 KHz voice spectrum at twice the frequency. This results in 8K samples per second. Each sample is 8 bits x 8K for 64 kbps. Digital voice is encoded using PCM (Pulse Code Modulation) ITU G.711.
Newer voice compression algorithms model PCM voice more efficiently using fewer bits to reduce the bandwidth required, while preserving the quality or audibility of the voice transmission. Low bit rate voice compression algorithms, such as ITU G.723.1 and G.729A permit the greatest number of simultaneous calls while maintaining high voice quality. In this way, compressed voice systems (CVS) can offer greater bandwidth savings, reduced network congestion and high quality voice transmissions.
Voice over IP (VoIP) is a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). Voice is sent in digital form in discrete packets over the IP network instead of in analog or digital form over a Time Division Multiplexing circuit over the public switched telephone network.
RAD offers a variety of product solutions supporting analog and digital voice, compressed voice and VoIP.
