P-2302R Series User’s Guide

 

Table 23 VoIP Advanced (continued)

 

 

 

 

LABEL

DESCRIPTION

 

 

 

 

Server Address

Your VoIP service provider must host a STUN server in order for you to use STUN.

 

 

Type the IP address or domain name (up to 127 ASCII characters) 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.

 

Use NAT

 

 

 

 

 

Active

Check this box to use a NAT router’s public IP address and SIP port number in the

 

 

Prestige’s SIP messages. This eliminates the need for STUN or a SIP ALG. You

 

 

must also configure the NAT router to forward traffic with this port number to the

 

 

Prestige.

 

Server Address

Enter the NAT router’s public IP address or domain name (up to 127 ASCII

 

 

characters) in this field.

 

Server Port

Enter the port number that your SIP sessions use with the public IP address of the

 

 

NAT router.

 

Outbound Proxy

 

 

 

 

 

Active

Check this box if your VoIP service provider has a SIP outbound server to handle

 

 

voice calls. This allows the Prestige to work with any type of NAT router and

 

 

eliminates the need for STUN or a SIP ALG. Turn off any SIP ALG on a NAT router

 

 

in front of the Prestige to keep it from retranslating the IP address (since this is

 

 

already handled by the outbound proxy server).

 

Server Address

Enter the IP address or domain name (up to 127 ASCII characters) of the SIP

 

 

outbound proxy server in this field.

 

Server Port

Enter the SIP outbound proxy server’s listening port for SIP outbound proxy

 

 

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 the SIP outbound proxy server.

 

NAT Keep Alive

 

 

 

 

 

Enable NAT Keep

You must have outbound proxy enabled to use NAT keep alive.

 

Alive

Enable NAT keep alive to have the Prestige send SIP notify messages to the SIP

 

 

server. Use this to keep a NAT router located between the Prestige and the SIP

 

 

server from timing out and dropping your Prestige’s SIP NAT sessions.

 

Keep Alive Interval

Set how often (in seconds) the Prestige should send SIP notify messages to the

 

 

SIP server.

 

Dual-Tone-Multi-

 

 

Frequency

 

 

(DTMF)

 

 

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.

 

 

 

 

MWI (Message

 

 

Waiting Indication)

 

Chapter 8 VoIP Screens

108