Section 4.4. Time Division Multiplexing

4.4. Time Division Multiplexing

Traditional PBX systems use a digital bus to carry sound information between interfaces where phones and/or trunks are connected. The signals flowing across this bus are digitized audio that travel in an aggregate formthat is, one bus can carry many separate signals within a single bit stream. The transmission technique is called time division multiplexing (TDM). To understand how all modern telephony solutions work, including VoIP, a basic understanding of TDM is important.

Multiplexing means combining many signals onto a single transport mechanism, such as a T1 or a PBX bus. This bus can be a connection between two pointslike a point-to-point T1 circuit, or it can be a large group of digital phones, like a PBX's bus.

Time division is the method of combining, and later dividing, the signals, with the purpose of yielding greater efficiency over the data link, be it a T1 circuit or a PBX backplane. Each signal is given a time slice, a small piece of the total bandwidth of the bus. At a very high rate, a TDM bus transmits a fixed sequence of time slices that are equal in duration. Each time slice contains a digitally sampled representation of the original analog waveform signal.

Each endpoint pulls a particular time slice out of the aggregate TDM bit stream and reassembles it, in real time, into a single cohesive digital signal. Each piece of the time slice used to reassemble that single signal is called a frame , just as pieces of the bit stream on an Ethernet data link or T1 are called frames .

4.4.1. Pulse Code Modulation and DS0 Channels

A sampling rate of about 8 KHz is sufficient to adequately record the human voice using analog instruments. When analog sound information is recorded at this rate, its amplitude, or power, is sampled 8,000 times per second. This is only the first step in putting the sound into digital format. It's still analog, because it is still measured on a scale with infinite, analog resolution. Imagine the sampled sound levels are 8,000 points on a line graph. Now draw curves to connect them. This is an analog waveform signal (see Figure 4-7).

In order to digitize that analog signal, it must be quantized, or put into a scale with limited resolution. Quantizing means taking each of those 8000 sample points and assigning each one to the nearest value on a finite, graduated scale. There are two major scales in useAlaw and m law. m law is prevalent in North America, Japan, and Hong Kong, while Alaw is common elsewhere. There are 256 levels on both scales. Eight thousand times per second, an analog sample's strength, or amplitude, is rounded to the closest of the 256 levels (see Figure 4-8).

Once digitally quantized, the signal must now be encoded. Since an 8-bit expression can describe 256 discrete values, and the rate of sampling is 8 KHz, the digital sound

Figure 4-7. An analog sample of part of a waveform signal

Figure 4-8. A digitized sample of the same part of the waveform signal in Figure 4-7

signal is transmitted at a data rate of 64 kbps, which is 8 bits times 8,000 samples per second. Lined up one after the other, 8,000 times per second, each 8-bit word forms a stream of binary data that travels the TDM bus in a single time slice. The common name for a 64 kbps voice channel is DS0 , and the signaling and encoding technique is called PCM, or pulse code modulation.

DS0 channels are the building blocks of ISDN and T1 voice circuits, too. Twenty-four of them together comprise a T1. This is why T1s have a maximum bandwidth of 1.5 Mbps, which is the same as 24 times 64 kbps.

4.4.2. Channel Banks

Channel banks are devices that multiplex (combine) separate digital voice signals into a single T1 circuit and demux (split) it back into its individual signals again. In a nutshell , a channel bank combines up to 24 separate phone lines into a single digital link.

Channel banks are not DSU/CSUs, which are data transmission devices, though some DSU/CSU devices can perform channel bank functions like connection of one or more single analog phone lines to a T1. For example, certain AdTran DSU/CSUs can hold FXO/FXS interface cards for connecting phones, PSTN lines, or PBXs. One purpose of a channel bank might be connecting a group of digital phones to a PBX through that PBX's T1 interface.

4.4.3. BRI

Basic Rate Interface (BRI) is an access signaling standard for multiplexing two voice channels on a single digital circuit. A third channel is used for call control. This is the service loosely referred to as an ISDN line, though ISDN itself means far more than just BRI. In fact, the PRI standard is also a child of ISDN.

BRI uses two channels of voice (called B channels), each occupying 64 kbps of bandwidth, while the third channel (called the D channel) occupies 16 kbps for call signaling. The B channels are purely for transmission of payloadthat's voice or data, while the third channel, the D channel, is used to signal connects and disconnects on the channel. Generally, digitizing for BRI is performed using a PCM algorithm, like a T1 or a digital TDM phone.

Switching to VoIP
Switching to VoIP
ISBN: 0596008686
EAN: 2147483647
Year: 2005
Pages: 172

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