How can I optimize the voice quality for VoIP connections?
Functions to improve the voice quality on VoIP connections are available on the Web configuration page: Settings - Telephony - Audio
The voice quality for VoIP connections is mainly determined by the voice codec used for transferring the data and the available bandwidth of your DSL connection.
In the case of the voice codec, the voice data is digitalised (coded/decoded) and compressed. A "better" codec (better voice quality) means more data needs to be transferred, i.e., perfect voice data transfer requires a DSL connection with a larger bandwidth.
You can influence the voice quality by selecting (bearing in mind the bandwidth of your DSL 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.
Default settings for the codecs used are stored in your phone; one setting optimised for low bandwidths and one for high bandwidths.
You can generally select one of these standard settings for all VoIP connections on your phone. If your DSL connection has a low bandwidth, you can also exclude parallel VoIP connection to increase the voice quality.
You can also make the settings for the voice codecs yourself by selecting the voice codecs to be used for each VoIP connection on your phone and specifying the sequence in which they should be suggested when establishing a VoIP connection.
The following voice codecs are supported by your phone:
Excellent voice quality. The broadband speech codec G.722 works at the same bit rate as G.711 (64 kbit/s per speech connection) but with a higher sampling rate. This allows higher frequencies to be played back. The speech tone is therefore clearer and better than for the other codecs (High Definition Sound Performance).
G.711 a law / G.711 μ law
Excellent voice quality (comparable with ISDN). The necessary bandwidth is 64 kbit/s per voice connection.
Good voice quality (inferior to that with G.711 but better than with G.729). Your phone supports G726 with a transmission rate of 32 kbit/s per voice connection.
Average voice quality. The necessary bandwidth is less than or equal to 8 kbit/s per voice connection. To save additional bandwidth and transmission capacity, on VoIP connections that use the G.729 codec you can suppress the transmission of voice packets in pauses ("Silence Suppression"). Instead of the background noises in your environment, your caller then hears a synthetic noise generated in the receiver (option: Enable Annex B for codec G.729).
Observe the following for good voice quality:
- When making calls using VoIP, avoid performing other Internet activities (e.g., surfing the Internet).
- Please note that voice delays can occur depending on the codec used and the network capacity utilisation.