. Core Network Planning and Optimisation Codec Codec lu - User Plane Nb - User Plane Physical Physical lu-CS Figure 4.15 User plane protocol stack Voice Codec on the User Plane Adaptive Multirate (AMR) Speech Codec The AMR codec was developed by the European Telecommunications Standards Institute (ETSI) and standardised for GSM; the codec has also been chosen by 3GPP as the mandatory codec for 3G systems. It is a multimode codec with eight narrow band modes with bit rates between 4.75 and 12.2 kbps. The sampling frequency is 8000 Hz and processing is done on 20 ms frames, i.e. 160 samples per frame. The codec was developed to preserve high speech quality under a wide range of transmission conditions. The AMR modes are closely related to each other and use the same coding framework. The codec is designed with a voice activity detector (VAD) and generation of comfort noise (CN) parameters during silence periods. The AMR codec, therefore, can reduce the number of transmitted bits and packets during silence periods to a minimum. The operation to send CN parameters at regular intervals during silence periods is usually called discontinuous transmission (DTX) or source controlled rate (SCR) operation. The format for packetisation of AMR encoded speech signals into the real-time transport protocol (RTP) is specified by RFC3267. The payload format supports transmission of multiple channels, multiple frames per payload, use of fast codec mode adaptation, robustness against packet loss and bit errors, and the inter-operation with existing AMR transport formats on non-IP networks. The speech bits encoded in each AMR frame have different perceptual sensitivities to bit errors. This property has been used in cellular systems to achieve better voice quality using unequal error protection and detection (UEP and UED) mechanisms. The UEP/UED mechanisms focus the protection and detection of corrupted bits onto the perceptually most sensitive bits in an AMR frame. In particular, speech bits in an AMR frame are divided into classes 