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Codecs are used to convert an analog voice signal to digitally encoded
version. Codecs vary in the sound quality, the bandwidth required, the
computational requirements, etc. You can specify which audio coding
process you would like to use. There are four voice codecs supported by
the VoIP router, you may try different settings to determine the best audio
quality you obtain from the combination of your network connection and
your used audio device (head set or hand set). The default codec sequence
is listed below. You can use the Up and Down buttons to change priority.
1. G.711 A law
2. G.711 U law
3. G.729
4. G.723.3
See the table below for a description of the parameters.
Parameter
Support Call
Waiting
Caller-ID
Presentation
Support User-Agent
Header
Support Out of
Band DTMF
Support Fake PSTN
Dial tone for VoIP
call
Support Freenet for
nonstandard VoIP
call flow
Support Freenet
IGNORE_ACK
Description
Enables or disables support for call waiting.
(Default: Disabled)
Check this check box to display the caller's number.
(Default: Disabled)
Enables or disables user-agent header support. Enabling this
feature includes user agent information in the packet, e.g., the
caller's ID may be displayed. (Default: Disabled)
Check this check box to support out of band Dual Tone Multi-
Frequency (DTMF). DTMF assigns a specific frequency
(consisting of two separate tones) to each key so that it can
easily be identified by a microprocessor. (Default: Disabled)
Check this check box to support fake PSTN dial tone for VoIP
calling. (Default: Disabled)
Check this check box to support Freenet for nonstandard
VoIP call flow. (Default: Disabled)
Check this check box to support Freenet IGNORE_ACK.
(Default: Disabled)
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IP
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