ZyXEL P-2002 User Guide - Page 62

The following table describes the labels in this screen.

Page 62 highlights

P-2002 Series User's Guide The following table describes the labels in this screen. Table 14 VoIP Advanced LABEL DESCRIPTION Advanced VoIP Settings This read-only field displays the number of the SIP account that you are configuring. The changes that you save in this page affect the Prestige's settings with the SIP account displayed here. SIP Server Settings URL Type Select SIP to have the Prestige include the domain name with the SIP number in the SIP messages that it sends. Select TEL to have the Prestige use the SIP number without a domain name in the SIP messages that it sends. Expiration Duration This field sets how long an entry remains registered with the SIP register server. After this time period expires, the SIP register server deletes the Prestige's entry from the database of registered SIP numbers. The register server can use a different time period. The Prestige sends another registration request after half of this configured time period has expired. Register Re-send Use this field to set how long the Prestige waits before sending a repeat Timer registration request if a registration attempt fails or there is no response from the registration server. Session Expires Use this field to set the longest time that the Prestige will allow a SIP session to remain idle (without traffic) before dropping it. Min-SE When two SIP devices negotiate a SIP session, they must negotiate a common expiration time for idle SIP sessions. This field sets the shortest expiration time that the Prestige will accept. The Prestige checks the session expiration values of incoming SIP INVITE requests against the minimum session expiration value that you configure here. If the session expiration of an incoming INVITE request is less than the value you configure here, the Prestige negotiates with the other SIP device to increase the session expiration value to match the Prestige's minimum session expiration value. RTP Port Range Real time Transport Protocol is used to handle voice data transfer. Use this field to configure the Prestige's listening port range for RTP traffic. Leave these fields set to the defaults if you were not given a range of RTP ports to use. Preferred Use this field to select the type of voice coder/decoder (codec) that you want the Compression Type Prestige to use. G.711 provides higher voice quality than G.729 but requires 64kbps of bandwidth while G.729 only requires 8kbps. Select G.711>G.729 if you want the Prestige to first attempt to use the G.711 codec and then the G.729 codec if the peer is not set up to use G.711. Select G.711 only if you want the Prestige to only use the G.711 codec when making VoIP calls. You will not be able to connect to a peer that is not set up to use G.711. Select G.729>G.711 if you want the Prestige to first attempt to use the G.729 codec and then the G.711 codec if the peer is not set up to use G.729. Select G.729 only if you want the Prestige to only use the G.729 codec when making VoIP calls. You will not be able to connect to a peer that is not set up to use G.729. STUN Use STUN if there is a NAT router between the Prestige and the voice service provider's SIP server. You do not need to use STUN if the NAT router is also a SIP ALG. Server Address Your VoIP service provider must host a STUN server in order for you to use STUN. Type the IP address of the STUN server in this field. 62 Chapter 6 VoIP Screens

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P-2002 Series User’s Guide
62
Chapter 6 VoIP Screens
The following table describes the labels in this screen.
Table 14
VoIP Advanced
LABEL
DESCRIPTION
Advanced VoIP
Settings
This read-only field displays the number of the SIP account that you are
configuring. The changes that you save in this page affect the Prestige’s settings
with the SIP account displayed here.
SIP Server
Settings
URL Type
Select
SIP
to have the Prestige include the domain name with the SIP number in
the SIP messages that it sends. Select
TEL
to have the Prestige use the SIP
number without a domain name in the SIP messages that it sends.
Expiration
Duration
This field sets how long an entry remains registered with the SIP register server.
After this time period expires, the SIP register server deletes the Prestige’s entry
from the database of registered SIP numbers. The register server can use a
different time period. The Prestige sends another registration request after half of
this configured time period has expired.
Register Re-send
Timer
Use this field to set how long the Prestige waits before sending a repeat
registration request if a registration attempt fails or there is no response from the
registration server.
Session Expires
Use this field to set the longest time that the Prestige will allow a SIP session to
remain idle (without traffic) before dropping it.
Min-SE
When two SIP devices negotiate a SIP session, they must negotiate a common
expiration time for idle SIP sessions. This field sets the shortest expiration time that
the Prestige will accept. The Prestige checks the session expiration values of
incoming SIP INVITE requests against the minimum session expiration value that
you configure here. If the session expiration of an incoming INVITE request is less
than the value you configure here, the Prestige negotiates with the other SIP
device to increase the session expiration value to match the Prestige’s minimum
session expiration value.
RTP Port Range
Real time Transport Protocol is used to handle voice data transfer. Use this field to
configure the Prestige’s listening port range for RTP traffic. Leave these fields set
to the defaults if you were not given a range of RTP ports to use.
Preferred
Compression Type
Use this field to select the type of voice coder/decoder (codec) that you want the
Prestige to use. G.711 provides higher voice quality than G.729 but requires
64kbps of bandwidth while G.729 only requires 8kbps.
Select
G.711>G.729
if you want the Prestige to first attempt to use the G.711 codec
and then the G.729 codec if the peer is not set up to use G.711.
Select
G.711 only
if you want the Prestige to only use the G.711 codec when
making VoIP calls. You will not be able to connect to a peer that is not set up to use
G.711.
Select
G.729>G.711
if you want the Prestige to first attempt to use the G.729 codec
and then the G.711 codec if the peer is not set up to use G.729.
Select
G.729 only
if you want the Prestige to only use the G.729 codec when
making VoIP calls. You will not be able to connect to a peer that is not set up to use
G.729.
STUN
Use STUN if there is a NAT router between the Prestige and the voice service
provider’s SIP server.
You do not need to use STUN if the NAT router is also a SIP ALG.
Server Address
Your VoIP service provider must host a STUN server in order for you to use STUN.
Type the IP address of the STUN server in this field.