Gigaset N670 IP PRO Installation, Configuration And Operation page 38

Provider and PBX profiles
Settings for codecs
The voice quality of VoIP calls is mainly determined by the codec used for the transmission and
the available bandwidth of your network connection. A "better" codec (better voice quality)
means more data needs to be transferred, i.e. it requires a network connection with a larger
bandwidth. You can change the voice quality by selecting 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. Default settings for the codecs used are stored in your phone; one setting optimised
for low bandwidths and one for high bandwidths.
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.
Active codecs / Available codecs
The following voice codecs are supported:
G.722 Outstanding voice quality. The G.722 wideband voice codec works at the same bit rate
as PCMA/PCMU (64 kbit/s per voice connection) but at a higher sampling rate (16 kHz).
To enable wideband connections via G.722 you have to activate the codec explicitly on
the Telephony – VoIP page (
PCMA/ (Pulse Code Modulation) Excellent voice quality (comparable with ISDN). The required
PCMU bandwidth is 64 kbit/s per voice connection.
PCMA (G.711 a law): Used in Europe and most countries outside of USA.
PCMU (G.711 μ law): Used in USA.
G.729A Average voice quality. The necessary bandwidth is less than or equal to 8 kbit/s per
voice connection.
Activate/deactivate a codec:
Select the required codec from the Available codecs/Active codecs list and click on
Define the sequence in which the codecs should be used:
In the Active codecs list select the required codec and click on
Selection of codecs G.722 and G.729 influence the system capacity in direction to
lower amount of parallel calls per base station.
Number of parallel calls per base station depending on codec
Codecs enabled
G729 and G711
G722 and G729 and G711
RTP Packetisation Time (ptime)
Length of time in milliseconds represented by the audio data in one packet.
Select the size of RTP packets to send. Select between 10 / 20 / 30 ms.
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Number of calls
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5
/ to move it up/down.
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