Avoid SIP NAT Traversal

This page has moved. See Asterisk Avoid SIP NAT Traversal for latest version


Using Asterisk as a SIP/IAX Gateway

The Asterisk open source telephony server can be used as a gateway in order to avoid SIP NAT Traversal. All VoIP devices on the LAN are configured to connect to an Asterisk server on the same LAN. The Asterisk server then connects to VoIP services on the Internet using the NAT friendly IAX protocol. The same technique can be applied to any other conventional VoIP protocol which has trouble traversing NAT.


On MacOSX, the SIP/IAX gateway setup for FWD as shown in the above diagram can be very easily configured using the Asterisk Assistants for MacOSX. It takes only a couple of minutes including installation and it requires no prior knowledge of Asterisk or VoIP.

Using the IAXy Analog Telephone Adapter

Alternatively, the IAXy Analog Telephone Adapter from Digium can be used in combination with any analog telephone to avoid SIP NAT Traversal. The analog telephone is connected to the IAXy using ordinary telephone wiring. The IAXy then connects to VoIP services on the Internet using the NAT friendly IAX protocol.


Potential problems

  • You will not be able to dial by SIP url with this configuration, i.e. dial sip:username@domain.tld - you will only be able to dial by FWD number.


This page has moved. See Asterisk Avoid SIP NAT Traversal for latest version


Using Asterisk as a SIP/IAX Gateway

The Asterisk open source telephony server can be used as a gateway in order to avoid SIP NAT Traversal. All VoIP devices on the LAN are configured to connect to an Asterisk server on the same LAN. The Asterisk server then connects to VoIP services on the Internet using the NAT friendly IAX protocol. The same technique can be applied to any other conventional VoIP protocol which has trouble traversing NAT.


On MacOSX, the SIP/IAX gateway setup for FWD as shown in the above diagram can be very easily configured using the Asterisk Assistants for MacOSX. It takes only a couple of minutes including installation and it requires no prior knowledge of Asterisk or VoIP.

Using the IAXy Analog Telephone Adapter

Alternatively, the IAXy Analog Telephone Adapter from Digium can be used in combination with any analog telephone to avoid SIP NAT Traversal. The analog telephone is connected to the IAXy using ordinary telephone wiring. The IAXy then connects to VoIP services on the Internet using the NAT friendly IAX protocol.


Potential problems

  • You will not be able to dial by SIP url with this configuration, i.e. dial sip:username@domain.tld - you will only be able to dial by FWD number.


Created by: benjk, Last modification: Wed 10 of Sep, 2008 (08:21 UTC) by admin
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