User`s guide

Chapter 11 The Service Configuration Screens
User’s Guide
117
Expiration
Duration
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.)
Register Re-send
timer
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.
Session Expires 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.
Min-SE 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.
RTP Port Range
Start Port
End Port
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.
Voice Compression
Primary,
Secondary, and
Third
Compression
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.7 11 A is typically used in Europe.
G.7 11 u is typically used in North America and Japan.
G.7 2 3 provides good voice quality, and requires 20 or 40 kbps.
G.7 2 9 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
Voice Coding on page 115
DTMF Mode
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
STUN
Active Select this if all of the following conditions are satisfied.
There is a NAT router between the WiMAX Modem and the SIP server.
The NAT router is not a SIP ALG.
Your VoIP service provider gave you an IP address or domain name for a
STUN server.
Otherwise, clear this field.
Server Address Enter the IP address or domain name of the STUN server provided by your VoIP
service provider.
Server Port Enter the STUN server’s listening port, if your VoIP service provider gave you one.
Otherwise, keep the default value.
Use NAT
Table 45 VOICE > Service Configuration > SIP Settings > Advanced (continued)
LABEL DESCRIPTION