G.711.1 speech codec was standardized by ITU-T in 2008. The codec operates on each 5 ms frame of 16-bit speech/audio signals sampled at 8 KHz or 16 KHz and generates a compressed bit-stream having bit-rates of 64 kbps, 80 kbps, or 96 kbps structured as 3 layers. The layered approach allows the decoder or any other component in the communication system to truncate the bit-stream by removing the higher layers. The base layer (Layer 0), at 64 kbps, is inter-operable with the traditional G.711. Layer 1 provides a lower band enhancement bit-stream at 16 kbps. Layer 2 provides a higher band enhancement bit-stream for the 16 KHz sampling frequency at 16 kbps. To improve the quality under frame erasures due to channel errors such as packet-losses, frame erasure concealment algorithms are provided at the decoder. This codec aims to achieve high-quality speech services over broadband networks, particularly for IP phone and multi-point speech conferencing, while enabling a seamless interoperability with conventional terminals and systems equipped only with G.711.