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senting seven channel sets The frequencies used for the channel sets in the A cells are the same, as are the frequencies in the B cells, C cells, etc There are many possible frequency reuse patterns Since the total number of channels for cellular and PCS is fixed, the selection of a reuse pattern and cell size determine how many subscribers can be supported in a given service area Available capacity, however, is much greater than the actual number of channels accessible in a given cell This is due to the nature of end-user calling behavior: not everyone wants to talk at the same time A cell can typically serve 10 to 20 times more subscribers than the total number of channels supported

Packet loss is a key voice impairment factor For voiceband connections, ITU-T G821 specifies the overall requirements for error performance in terms of errored seconds and severely errored seconds Under this definition, for the majority of voice-encoding schemes, the loss of a single VoIPoMPLS packet will cause at least a single severely errored second ITU-T G821 specifies an end-to-end severely errored seconds (SESs) requirement of 1 in 10-3 this requirement is predominately driven by the demands of voiceband data (fax and modem) Speech impairment in packetized voice networks, on the other hand, can be unnoticeable with fairly high packet loss (as high as 5 percent in some cases) The relationship between SES and packet loss is not well known In networks where it is important to pass voice, modem, and/or fax data without degradation, techniques such as controlling packet loss may be employed Alternatively, demodulation, data pass-through, and remodulation of fax/modem calls may be employed to achieve such a goal

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Digital radio technologies have been developed for use on the air (or radio) interface These can dramatically increase the number of subscribers supported on the range of frequencies used for wireless telecommunications systems (note that the analog NAMPS technology also provides an increase in the number of subscribers supported) The two basic types of digital technology are time division multiple access (TDMA) and code division multiple access (CDMA) Many standards exist for the use of these basic technologies TDMA is based on the use of time-interleaving of multiple signals to provide an apparently simultaneous transmis-

Downloaded from Digital Engineering Library @ McGraw-Hill (wwwdigitalengineeringlibrarycom) Copyright 2004 The McGraw-Hill Companies All rights reserved Any use is subject to the Terms of Use as given at the website

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Part 1:

sion of those signals on a single radio frequency CDMA is based on a technique known as direct sequence spread spectrum (DSSS) that digitally codes a base signal and employs different signal encoding patterns and frequency hopping to redistribute that base signal across a broad range of frequencies Logical channels for each signal are created through the use of unique code sequences TDMA and CDMA can provide many advantages over analog-based systems Examples are better voice quality, increased capacity, less noise and interference, and the ability to provide digital services such as data and messaging The ANSI-41 networking protocol is designed to support versions of these newer-generation digital technologies as well as the original analog systems

Part 1:

When determining the timing accuracy for VoIPoMPLS domains, the following types of traffic must be considered: speech, voiceband data, and circuit-mode data All speech traffic is obtained by the equivalent of sampling the analog speech signal at a nominal 8 kHz and generating linear PCM This can be companded to 64 Kbps in accordance with ITU-T G711, or it can be compressed to a lower bit rate either on a sample-by-sample basis (such as ADPCM G726/7) or on a multiple-sample basis to produce packets (such as various forms of code-excited linear prediction [CELP]) Voiceband data traffic is obtained by sampling the analog modem signal, that is, the low-rate data modulated onto defined frequency carrier signals, in the same way as for speech and companding to 64 Kbps using G711 Except for very low data rates, compression is not possible In all cases, provided the traffic can be carried by the VoIPoMPLS packet network directly from encoder to decoder and the decoder can work on the sample rate determined from the received traffic, the encoder only needs to have a frequency tolerance sufficient to achieve the required analog frequency response and constrain the traffic data bandwidth; thus, the VoIPoMPLS packet network would have no particular frequency tolerance requirements (Packet jitter including delay variation would still have to be constrained within buffer sizes, and measures such as sequence numbers would still be needed to maintain accurate determination of the transmitter sample rate under circumstances of packet loss)

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