Voice Quality - Aastra OpenCom 510 User Manual

Voice over IP (VoIP)
Required bandwidth (kilobit/s) with respect to Packet Length and Codec
Packet G.711 (not
Length (ms) compressed)
60
70
80

10.1.3 Voice Quality

The achievable voice quality depends on various factors. It is possible to optimise
voice-data transmission on an existing network using the available configuration
settings. Measuring the network quality may also help.
The following comparison provides a guide to voice quality with specific quality
levels:
Quality Levels for Voice Transmission with VoIP
Level Voice Comprehensibility
1 Very Good
2 Good
3 Satisfactory
4 Limited
> 4 Unacceptable
When a call is set up, the terminals involved negotiate the voice-data compression
("codec") that will be used. This is the first factor that determines the achievable
quality level:
■
G.711 A-Law (Level 1, uncompressed): The audio data of a PCM channel
(64 kilobit/s) is adopted one-to-one. Every VoIP terminal must support this
codec. This codec can not be used with an ISDN data connection.
114
G.729A
c. 6.3 kilobit/s
33.6
31.09
29.2
Note: To ensure SIP compatiblility, the older system telepho-
nes OpenPhone 63 IP and OpenPhone 65 IP does not sup-
port the G.723 codec any more.
G.723.1
6.3 kilobit/s
30.4
Comparable to
ISDN
DECT
GSM
Defective GSM
No Connection
Fundamentals
G.723.1
5.3 kilobit/s
28.3