Voip Introductory Technical Description

Figure 15.3 shows a simplified block diagram of VoIP operation from an analog signal deriving from a standard telephone, which is digitized and transmitted over the Internet via a conversion device. Then, at the distant end, it is converted back to analog telephony using a similar device suitable for input to a standard telephone. The gateway is placed between the voice codec and the digital data transport circuit. An identical device will also be found at the far end of the link. This equipment carries out the signaling role on a telephone call among other functions.

Moving from left to right in Figure 15.3, we have the spurty analog signal deriving from a standard telephone set. The signal is then converted to a digital counterpart using one of seven or so codecs [coder-decoder(s)] that the VoIP system designer has to select from. Some of the more popular codecs for this application are listed in Table 15.1. The binary output of the codec is then applied to a conversion device (i.e., a "packetizer") that loads these binary 1s and 0s into an IP payload of from 20 to 40 octets in length.

The output of the converter consists of IP packets1 that are transmitted on the web or other data circuit for delivery to the distant end.

At the far end the IP packets or frames are input to a converter (i.e., depacketizer) that strips off the IP header, stores the payload, and then releases it in a constant bit stream to a codec (i.e., a D-A converter). Of course this codec must be compatible with its

Figure 15.3 Elements of basic operation of VoIP where the input signal derives from a conventional analog telephone. This figure follows Figure 15.1 in sequence of complexity.

Figure 15.3 Elements of basic operation of VoIP where the input signal derives from a conventional analog telephone. This figure follows Figure 15.1 in sequence of complexity.

1 The output may be ATM cells (see Chapter 20) if the intervening network is an ATM network.

Table 15.1 Characteristics of Speech Codecs Used in Packet Networks

Voice Bit

Voice Frame

Header

Packets per

Packet Bit

Coding Algorithm

Rate (kbits/sec)

Size (bytes)

(bytes)

Second

Rate (kbits/sec)

G.711 8-bit PCM [1]

64

80

40

100

96

G.723.1 MPMLQa [4]

6.3

30

40

26

14.6

G.723.1 ACELPb [5]

5.3

30

40

22

12.3

G.726 ADPCMC [6]

32

40

40

100

64

G.728 LD-CELP0 [7]

16

20

40

100

48

G.729a CS-ACELP® [8]

8

10

40

100

40

aMPMLQ, multipulse maximum likelihood quantization.

bACELP, algebraic code-excited linear prediction.

°ADPCM, adaptive differential PCM.

dLD-CELP, low delay code-excited linear prediction.

eCS-ACELP, conjugate structure algebraic code-excited linear prediction.

aMPMLQ, multipulse maximum likelihood quantization.

bACELP, algebraic code-excited linear prediction.

°ADPCM, adaptive differential PCM.

dLD-CELP, low delay code-excited linear prediction.

eCS-ACELP, conjugate structure algebraic code-excited linear prediction.

near-end counterpart. The codec converts the digital bit stream back to an analog signal that is then input to a standard telephone subset.

The insightful reader will comment that many steps of translation and interface have been left out. Most of these considerations will be covered below in Section 15.4.1 in our discussion of the gateway.

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