The following table describes the labels in this screen.
Table 14 VoIP Advanced
LABEL | DESCRIPTION |
|
|
Advanced VoIP | This |
Settings | 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 | This field sets how long an entry remains registered with the SIP register server. |
Duration | 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 | 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. |
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 |