User's Manual

SME VoIP System Guide, Version 1.5
Proprietary and Confidential
Chapter: Core Network Server(s) Configuration
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8.6 SIP Server Setup
SIP server is one of the main components of an SME network, dealing with the setup of all SIP calls in the
network. A SIP server is also referred to as a SIP Proxy or a Registrar.
Although the SIP server is the most important part of the SIP based phone system, some servers only
handles call setup and call tear down. It does not actually transmit or receive any audio. This is done by the
media server in RTP.
The RTX SME family of network phones are fully interoperable with the most of SIP Server applications.
There are many off-the-shelf vendor and open source SIP servers. In this section, we will briefly explain
settings required to take full advantage of FreePBX SIP Server feature set. The settings are similar for other
SIP servers.
8.6.1 FreePBX SIP Server
FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk, which the most
popular open source telephony engine software.
The administrator should refer to the relevant detail step-by-step procedure of how to install FreePBX SIP
server. This section briefly describes SIP Server setup parameters.
1) SIP Server Setup
Settings
Description
NAT
This option determines the settings for users connecting to an asterisk server.
Possible values: Yes, No, Never, Route
NAT=route
Asterisk will send the audio to the port and IP where its receiving the audio from.
Instead of relying on the addresses in the SIP and SDP messages.
This will only work if the phone behind NAT send and receive audio on the same
port and if they send and receive the signalling on the same port. (The signalling
port does not have to be the same as the RTP audio port).
NAT=No
Asterisk will add an RPORT to the via header of the SIP messages
NAT=never
This will cause asterisk not to add an RPORT in the VIA line of the sip invite