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Chapter 1 Introduction 27

IP Telephony Configuration Guide

Public Switched Telephone Network

The Public Switched Telephone Network (PSTN) can play an important role in IP telephony

communications. In many installations, the PSTN forms a fallback route. If a call across a VoIP

trunk does not have adequate voice quality, the call can be routed across PSTN lines instead, either

on public lines or on a dedicated ISDN connection between the two systems (private network).

The Business Communications Manager also serves as a gateway to the PSTN for all voice traffic

on the system.

Key IP telephony concepts

In traditional telephony, the voice path between two telephones is circuit switched. This means

that the analog or digital connection between the two telephones is dedicated to the call. The voice

quality is usually excellent, since there is no other signal to interfere.

In IP telephony, each IP telephone encodes the speech at the handset microphone into small data

packets called frames. The system sends the frames across the IP network to the other telephone,

where the frames are decoded and played at the handset receiver. If some of the frames get lost

while in transit, or are delayed too long, the receiving telephone experiences poor voice quality.

On a properly-configured network, voice quality should be consistent for all IP calls.

The information under the following headings describe some of the components that determine

voice quality for IP telephones and trunks:

• “Codecs” on page 27

• “Jitter Buffer” on page 28

• “QoS routing” on page 29

Codecs

The algorithm used to compress and decompress voice is embedded in a software entity called a

codec (COde-DECode).

Two popular Codecs are G.711 and G.729. The G.711 Codec samples voice at 64 kilobits per

second (kbps) while G.729 samples at a far lower rate of 8 kbps. For actual bandwidth

requirements, refer to “Determining the bandwidth requirements” on page 153, where you will

note that the actual kbps requirements are slightly higher than label suggests.

Voice quality is better when using a G.711 CODEC, but more network bandwidth is used to

exchange the voice frames between the telephones.

If you experience poor voice quality, and suspect it is due to heavy network traffic, you can get

better voice quality by configuring the IP telephone to use a G.729 CODEC.

Note: You can only change the codec on a configured IP telephone if it is online to the Business

Communications Manager, or if Keep DN Alive is enabled for an offline telephone.

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Avaya IP Telephony Guía de configuración Pagina 27