Prophecy 2006

In build v.25, SIP Phone dndn't work correctly with Network Address Translation. As of build v.42 and v.48, it does work correctly, but you must configure your firewall correctly.

Goal of this section: to configure your firewall correctly and allow both your server and your SIP Phone to connect to outside SIP services, such as FWD or Vonage.

Background: To get SIP through a NAT firewall, you must either configure the firewall to let SIP through, or you must use a STUN server, or you must include tricks in your SIP Phone. P2006 (v.42+) does not support STUN, so here's how to configure your firewall.

Note: The P2006 Management Console (MC) asks for all the right variables to configure a tricky, firewall-traversing SIP Phone — but the server doesn't actually support it.

Note: There's a bug in V.25 of SIP Phone; regardless of firewall configuration, it won't traverse the firewall. That problem is fixed in v.42+.

The goal here is to have a configuration which allows the following call path:

SIP Phone -> Voxeo server -> FWD (or Vonage, etc.)

This firewall configuration information will look very familiar to anyone who has worked with SIP Phones before.

To handle the SIP Phone itself: by default, the SIP phone uses port 15000 as the "First RTP Port" for sending audio back and forth. Since the SIP phone has 3 lines, each of which can take 4 ports, ports 15000-15011 need to be forwarded to the PC that's running the SIP phone. I forward both TCP and UDP.

If the SIP Phone is to contact the outside world directly through the firewall, then add "NAT IP" translations.

To handle the Voxeo server: port 5060 and some higher ports need to be forwarded to the PC running the server. I use 5060 through 5072, both TCP and UDP.