Digital Subscriber Signaling System No. 1 DSS1

THE AAL 2 SERVICE-SPECIFIC CONVERGENCE SUBLAYER 307 to the destination User, which is responsible for decoding them into a sequence of audio samples. The following are some of the ITU-T audio algorithms. • G.711 pulse code modulation PCM : Produces one 8-bit value every 125 µsec, repre- senting the sign and amplitude of an audio sample. Two encoding laws – the A-law and the µ-law – can be used. It normally transmits at 64 Kbps, but it can also transmit at 56 Kbps and 48 Kbps. The G.711 output is accumulated over 1 msec to form an EDU of 8 encoded values. • G.722 sub-band adaptive pulse code modulation SB-ADPCM : Produces one 8-bit value every 125 µsec, and represents audio samples with higher fidelity than G.711 PCM. The EDU consists of eight encoded values i.e. values collected over 1 msec. • G.723.1 : Operates at either 5.3 or 6.4 Kbps. Both rates are a mandatory part of the encoder and decoder. Every 30 ms, it emits either 160 or 192 bits, respectively; this characterizes a voice sample. It is possible to switch between the two rates at any 30 msec boundary. • G.726 adaptive pulse code modulation ADPCM : Supports bit rates of 40, 32, 24, and 16 Kbps. Every 125 µsec, the encoding produces 5, 4, 3, or 2 bits, respectively. • G.722 embedded adaptive pulse code modulation EADPCM : This is a family of vari- able bit rate coding algorithms with the capability of bit dropping outside the encoder and decoder blocks. It produces code words which contain enhancement bits and core bits . The enhancement bits can be discarded during network congestion. The number of code bits must remain the same to avoid mistracking of the adaptation state between transmitter and receiver. Algorithms of the G.727 family are referred to by the pair x, y, where x is the number of core plus enhancement bits and y is the number of core bits. Recommendation G.727 provides coding rates of 40, 32, 24, and 16 Kbps, with core rates of 16, 24, and 32 Kbps. This corresponds to the following x, y pairs: 5, 2, 4, 2, 3, 2, 2, 2, 5, 3, 4, 3, 3, 3, 5, 4, and 4, 4. The data unit format requires that G.727 outputs be accumulated over an interval of 1 msec to yield a sequence of eight encoded values. • G.728 low delay code excited linear prediction LD-CELP : This coder produces a group of four codewords every 2.5 msec. Each group of codewords is referred to as an adaptation cycle or frame. • G.729 : Runs at 8 Kbps. Every 10 msec, it emits 80 bits that encode a voice frame. Multi-frequency tones and CAS bits At the transmitter’s side, it detects and extracts dialed digits codes from multi-frequency tones, such as DTMF. It also extracts the ABCD CAS bits. At the receiver’s side, it regenerates the multi-frequency tones from the received dialed digit codes and regenerates the ABCD CAS bits. Facsimile At the transmitter’s side, when it detects a facsimile, it demodulates the facsimile signal and sends the demodulated original image data and associated control signals to the transmitting SSCS. At the receiver’s side, it receives the image data and control signals from the receiving SSCS, it remodulates them into voiceband for transmission to the peer facsimile terminal. This demodulationremodulation procedure provides a higher-fidelity transfer. 308 VOICE OVER ATM AND MPLS Circuit mode data At the receiver’s side, it passes through circuit mode data, such as N × 64 Kbps fractional T1E1, and at the receiver’s side it regenerates the circuit mode data. Data frames At the transmitter’s side, it extracts payloads from data frames, and removes flags, bit stuffing and CRC. At the receiver’s side, it regenerates data frames and restores flags, bit stuffing, and CRC.

12.5.2 The Service-Specific Convergence Sublayer

Voice is real-time traffic, that has to be delivered to the destination with minimum jitter. Annoying distortions can result due to jitter variability and brief silence periods can be shortened or lengthened. Modem traffic also has to be delivered with minimum jitter, because abnormal phase shifts can be sensed if the delay varies. An SSCS transmitter passes information from its User to CPS with no delay variation. However, cell delay variation can be introduced at the CPS transmitter during periods of time when voice from too many AAL 2 connections is directed simultaneously onto the same ATM connection. Cell delay variation can be controlled by CAC and by requesting the user to switch to an algorithm with greater compression. As in the case of the AAL 1 convergence sublayer, the receiving SSCS introduces a delay before it delivers the information to the receiving User in order to cancel any delay variations incurred by the network. Type 1 and Type 3 packets A transmitting SSCS passes data to CPS in the form of a packet known as CPS-packet see Section 3.7.2. The structure of the CPS-packet is shown in Figure 3.20. It consists of a 3-byte header and a payload which has a maximum length of 45 bytes. In the AAL 2 SSCS for trunking, the CPS-packet payload has been further defined. Specifically, it can be either a Type 1 packet unprotected or a Type 3 packet fully protected. Type 2 packets are to be defined. In Type 1 packet, the CPS-packet payload is simply made up of data without any additional information that can be used for error detection, such as CRC or parity check. The maximum payload is 45 bytes. The format of the Type 3 packet is shown in Figure 12.13. The maximum payload is 43 bytes, and the remaining 2 bytes are used for the fields: message type and CRC. The message type is a 6-bit field and it contains a code to indicate the contents of the payload. Message type codes have been defined for dialed digits, ABCD CAS bits, facsimile demodulation control data, alarms, and user state control operations. The CRC-10 is a 10-bit field that contains the FCS computed using the polynomial x 10 + x 9 + x 5 + x 4 + x + 1. payload Message type CRC-10 Figure 12.13 Type 3 packet fully protected.