ZyXEL Communications MAX-306M1 User Manual page 138

Chapter 12 The Service Configuration Screens
Table 44 VOICE > Service Configuration > SIP Settings > Advanced (continued)
LABEL
Expiration
Duration
Register Re-
send timer
Session Expires
Min-SE
RTP Port Range
Start Port
End Port
Voice Compression
Primary,
Secondary, and
Third
Compression
DTMF Mode
138
DESCRIPTION
Enter the number of seconds your SIP account is registered with the
SIP register server before it is deleted. The WiMAX Modem
automatically tries to re-register your SIP account when one-half of this
time has passed. (The SIP register server might have a different
expiration.)
Enter the number of seconds the WiMAX Modem waits before it tries
again to register the SIP account, if the first try failed or if there is no
response.
Enter the number of seconds the conversation can last before the call is
automatically disconnected. Usually, when one-half of this time has
passed, the WiMAX Modem or the other party updates this timer to
prevent this from happening.
Enter the minimum number of seconds the WiMAX Modem accepts for a
session expiration time when it receives a request to start a SIP
session. If the request has a shorter time, the WiMAX Modem rejects it.
Enter the listening port number(s) for RTP traffic, if your VoIP service
provider gave you this information. Otherwise, keep the default values.
To enter one port number, enter the port number in the Start Port and
End Port fields.
To enter a range of ports:
• Type the port number at the beginning of the range in the Start
Port field
• Type the port number at the end of the range in the End Port field.
Select the type of voice coder/decoder (codec) that you want the
WiMAX Modem to use.
G.711 provides high voice quality but requires more bandwidth (64
kbps).
• G.711A is typically used in Europe.
• G.711u is typically used in North America and Japan.
• G.723 provides good voice quality, and requires 20 or 40 kbps.
• G.729 requires only 8 kbps.
The WiMAX Modem must use the same codec as the peer. When two
SIP devices start a SIP session, they must agree on a codec.
For more on voice compression, see
Control how the WiMAX Modem handles the tones that your telephone
makes when you push its buttons. You should use the same mode your
VoIP service provider uses.
• RFC 2833 - send the DTMF tones in RTP packets
• PCM - send the DTMF tones in the voice data stream. This method
works best when you are using a codec that does not use
compression (like G.711). Codecs that use compression (like G.729)
can distort the tones.
• SIP INFO - send the DTMF tones in SIP messages
Voice Coding on page 135
User's Guide