Prestige 2002 Series User’s Guide

Table 11 VoIP Advanced (continued)

LABEL

DESCRIPTION

 

 

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

Prestige to use. G.711 provides higher voice quality than G.729 but requires

Type

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

 

 

 

Active

Select this check box to turn on STUN. Use STUN if there is a NAT router between

 

the Prestige and the VoIP 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.

Server Port

Enter the STUN server’s listening port for STUN requests in this field. Leave this

 

field set to the default if your VoIP service provider did not give you a server port

 

number for STUN.

DTMF Mode

The Dual Tone Multi-Frequency (DTMF) mode sets how the Prestige handles the

 

tones that your telephone makes when you push its buttons. It is recommended

 

that you use the same mode that your VoIP service provider uses.

 

Select RFC 2833 to send the DTMF tones in RTP packets.

 

Select PCM (Pulse Code Modulation) to include 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) could distort

 

the tones.

 

Select SIP INFO to send the DTMF tones in SIP messages.

 

 

Back

Click Back to return to the VoIP screen without saving configuration changes.

 

 

Apply

Click Apply to save your changes back to the Prestige.

 

 

Reset

Click Reset to begin configuring this screen afresh.

 

 

50

Chapter 6 VoIP Screens