CCNA Voice 03

Analog circuits provide enough bandwidth to enable users to make a single call per line. This, however, is not scalable. Digital circuits makes use of a single line to carry multiple calls through multiplexing.
One particular multiplexing method used by many technologies is the Time Division Multiplexing, otherwise known as TDM. TDM divides time into many slots and allocates each call a fair slice of time for transmission over a single line. The receiving end then demultiplexes the incoming data into separate lines.

An example of a technology reliant on TDM is the T1 line. A T1 line bundles 24 voice channels into a single digital circuit. Each voice channel, known as DS0, provides 64Kbps of bandwidth. Combined, you'll have a speed of 1.536Mbps for data. However, framing the data requires an additional 8Kbps, which results in a raw speed of 1.544Mbps.

T1 is framed through two different formats:
1) D4 is the original format where 8 bits are used to frame each of the 24 DS0s, resulting in 192 bits of framing. An additional 1 bit is used for termination, resulting in 193 bits. 12 groups of such frames results in a SF (SuperFrame)

2) ESF is the modern T1 framing technique which extends the 12 frames to 24 frames.

D4 framing circuits are configured to have AMI line coding, while B8ZS line coding is used on ESF circuits.

Similar to a T1, an E1 line is commonly found outside North America. It is the European version of the T1 circuit. E1 circuits make use of Channel Associated Signaling (CAS) which robs the lower order bit of the sixth time slice for signaling. E1 makes use of 32 DS0s, where the first and seventeenth channel is used for framing and signaling respectively. The raw bandwidth is 2.048Mbps.

E1 CAS uses R2 signaling to exchange caller information. Three methods to exchange are:
1) R2-compelled - When the tone pair is sent from the switch, the tone stays on until the receiving end replies with a tone pair.
2) R2-noncompelled - When the tone pair is sent from the switch, it is sent in pulses. The replies are also sent in pulses. This is the most common implementation.
3) R2-semicompelled - When the tone pair is sent from the switch, the tone stays on while the replies are sent in pulses.

Like E1, the T1 circuit described thus far is the CAS variant, which also uses robbed bits. Another type of T1/E1 circuit is the ISDN circuit, which uses CCS (Common Channel Signaling). CCS uses an entirely separate channel to provide signaling and this signaling is done through HDLC at Layer 1, Q.921 at Layer 2 and Q.931 and Layer 3. Two types of CCS circuits are PRI and BRI, which stands for Primary Rate Interface and Basic Rate Interface respectively.

PRI provides 23 channels for data, and 1 channel for signaling, which are referred to as 23B+D. The E1 PRI is common in European countries and is referred to as 30B+D. BRI on the other hand provides 2B+D, resulting in 144Kbps (2 DS0 + a 16Kbps signaling channel).

Before voice can be transmitted over the network, you would require a process to convert an analog sound wave into a digital representation of it. The sampling rate for digitizing analog signals is determined using Nyquist's theorem which states that the sampling (baud) rate should be equal to twice or more of the original sample. The human voice scales from 300Hz to 3300Hz, so it makes good sense to sample at 8000Hz to be on the safe side.

The next process is quantization. Quantization uses a logarithmic scale to quantize the samples. Using such a scale provides greater granularity for smaller signals which allows a high signal-to-noise ratio.

Encoding turns the quantized values into binary. Currently there are two encoding standards in use: mu-law and A-law. mu-law is used in North America while A-law is used everywhere else. The difference between them is that mu-law actually inverts the segment and interval before creating the binary value, while A-law simply takes the values as it is.

Finally comes compression. The world's standard for communication is the G.711 codec, which is a direct translation of an analog signal to a voice which takes up 64Kbps of bandwidth. In order to increase the number of calls a DS0 can carry, compression is used. Compression algorithms such as ADPCM details the change in the wave rather than the entire wave. Prediction can also further help reduce the variations between the original wave and the new wave. Another compression algorithm known as CS-ACELP is also widely adopted. Algorithms like G.729 make use of CS-ACELP technology.