Siemens 2000 - Manual (Page 31)

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Audio Settings
Navigation: Admin > Audio Settings
The values for the voice quality are set at the factory to ensure the voice
quality is generally acceptable with the minimum possible use of resources
(bandwidth). You should therefore only change the preset values if the
voice quality seems to you to be too poor or if you wish to reduce the
bandwidth required at the expense of the voice quality (e.g. for parallel
connections to the Internet). Increasing the voice quality is usually also
associated with an increase in the bandwidth required.
The voice quality for VoIP is largely influenced by the execution time for the
voice packets.
The execution time determines the delay between the sender speaking and
the recipient receiving what has been said. It is composed of the following
parts:
• Time the used Codec needs to digitise the voice, packetise it in data
packets and, if necessary, to compress the packets,
• Time the voice packets spend in the Internet, essentially consisting of
the time the voice packets wait in the buffer of the IP node when there
is heavy traffic (see also Æ Quality of Service(QoS): Protocol Settings).
You can make the following settings:
Codec
Codec is a procedure used to digitise and packetise (if necessary also
compress ) the analogue voice before sending, and to decode the digital
data on receipt, i.e. translate it into analogue voice. The choice of Codec is
a compromise between voice quality, transmission speed, and the
necessary bandwidth.
Both parties involved in the telephone connection (caller/sender and
recipient) must use the same Codec. The Codec is negotiated between the
sender and the recipient when establishing a connection.
Select from the Codec list the Codec that the handset is to suggest when
establishing a connection.
You can choose between the following Codecs supported by your handset:
– G.711 preferred (normal quality)
The voice quality is very good, roughly corresponding to that of an
ISDN fixed network connection. The necessary bandwidth is 64 Kbit/
s per voice connection. As there is little compression, the delay
caused by coding/decoding is only 0.125 ms.
– G.723 preferred (low bandwidth) /
G.723 only (low bandwidth)
The voice quality is below that in mobile phone networks. The neces-
sary bandwidth is 24 or 40 Kbit/s per voice connection. The delay is
about 30 ms.