Switching to VoIP

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Copyright © 2007 O’Reilly & Associates, Inc. All rights reserved.

SIP registrations don’t work through a firewall

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registrar. The local socket used to communicate with the recipient is the local IP

address and port used by your phone’s RTP agent: i.e., your phone’s IP address and

RTP port. Consider Figure 15-1.

In Figure 15-1, phone A is able to send audio data to phone C, but phone C is unable

to send audio data to phone A because the firewall (B) doesn’t know what to do with

the RTP packets coming from phone C. This is a problem created by NAT (network

address translation), which cannot keep track of connectionless applications that use

more than one socket pair, like a two-way phone conversation.

To solve this problem, add a SIP proxy server between phone A and firewall B. Con-

figure the SIP phone so that it places all calls through the SIP proxy. The SIP proxy

knows how to handle the RTP data sent to, and received from, many SIP endpoints

simultaneously. In this instance, the SIP proxy would need to reside on a DMZ

between A and B, so that it can have a publicly routable IP address. The SIP proxy

could also reside on the same host as the firewall, if public IP addresses aren’t

abundant.

Another way to approach the NAT problem is to use a STUN (Simple Traversal of

UDP) NAT server. This server assists endpoint devices in figuring out what sockets ...