User manual

Maxwell 10 / en / A31008-N4001-R102-1-20 / web_configurator.fm / 22.01.2014

Telephony

Template A4, Version 1, 03.04.2012

Audio settings

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 simul

taneously – so that the transmission no longer takes place smoothly. The following settings allow you

to adjust your Gigaset to your individual DSL connection.

Settings Telephony Audio

You can set the following parameters for the voice quality for each connection:

Time interval for RTP packets

¤ 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 audio-

visual data (streams) via IP-based networks.

Voice quality

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 con

nection. 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.

¤ Select the required codecs and define the sequence in which they should be used.

The following voice codecs are supported by your phone:

Silence suppression

Silence suppression means that no data packets are sent during a pause in speaking. This means a

lower data volume but call participants may interpret it as an interruption to the connection.

¤ Select No if you do not want silence suppression. Default setting: Ye s

Saving settings

¤ Click on Save to save your settings on the screen.

G.722 The broadband voice codec G.722 works at the same bit rate as G.711 (64 kbit/s per

voice connection) but at a higher sampling rate (16 kHz) and therefore provides

excellent sound quality.

G.711 a law/G.711 μ law

Excellent voice quality (comparable with ISDN). The required bandwidth is 64 kbit/s

per voice connection.