Audio - Gigaset DE 900 IP PRO User Manual

Configuring phone settings via the Web configurator
Voice quality (audio)
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.
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Settings Telephony
You can set the following parameters for the voice quality:
Time interval for RTP packets
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Select the interval for sending RTP packets (20 or 30 ms).
RTP (RTP = Real-Time Transport Protocol) is a protocol for the continuous transmission of
audiovisual data (streams) via IP-based networks.
Voice quality
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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 estab-
lishing 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 connection is established.
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Select the required codecs and define the sequence in which they should be used.
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Audio