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Siemens gigaset 900 for 910 Manual

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Configuring phone settings via the Web configurator
Gigaset DE900 IP PRO / en / A31008-M2210-R101-3x-7619 / web_configurator.fm / 22.03.2011
Version 2, 27.05.2010
Voice quality
The voice quality of your VoIP calls is determined by the codec used for the transmission.
To increase the quality, more data must be transmitted. Depending on the bandwidth of
your DSL connection, this can then lead to problems with the volume of data – especially
if two VoIP calls are made simultaneously – so that the transmission no longer takes place
smoothly. The following settings allow you to adjust your Gigaset to your individual DSL
connection.
Settings
¢ Te lephony ¢ Voice Quality
You can set the following parameters for the voice quality:
Packed time for RTP packages
¤ Select the interval for sending RTP packets (20 or 30 ms).
RTP (Real-time Transport Protocol) is a protocol for the continuous transmission of
audiovisual data (streams) via IP-based networks.
Voice quality
¤ Select the voice quality that matches the bandwidth of your DSL connection.
– Optimised for high bandwidth
– Optimised for low bandwidth
–Select Own Codec preference if you want to determine the codecs yourself.
Both parties involved in a phone connection (caller/sender and recipient) must use the
same voice codec. The voice codec is negotiated between the sender and the recipient
when establishing a connection. You can influence the voice quality by selecting (bearing
in mind the bandwidth of your Internet connection) the voice codecs your phone is to
use, and specifying the order in which the codecs are to be suggested when a VoIP con-
nection is established.
¤ Select the required codecs and define the sequence in which they should be used.
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