Here we can show some examples of working configuration for Asterisk's SIP channel driver when Asterisk is behind NAT (Network Address Translation).

Asterisk and Phones Connecting Through NAT to an ITSP

This example should apply for most simple NAT scenarios that meet the following criteria:

• Asterisk and the phones are on a private network.
• There is a router interfacing the private and public networks. Where the public network is the Internet.
• The router is performing Network Address Translation and Firewall functions.
• The router is configured for port-forwarding, where it is mapping the necessary ranges of SIP and RTP traffic to your internal Asterisk server.
  In this example the router is port-forwarding WAN inbound TCP/UDP 5060 and UDP 10000-20000 to LAN 192.0.2.10

This example was based on a configuration for the ITSP SIP.US and assuming you swap out the addresses and credentials for real ones, it should work for a SIP.US SIP account.

Devices Involved in the Example

192.0.2.20

192.0.2.10

LAN: 192.0.2.1
WAN: 198.51.100.5

203.0.113.1(gw1.example.com)
203.0.113.2(gw2.example.com)

For the sake of a complete example and clarity, in this example we use the following fake details:

ITSP Account number:  1112223333

DID number provided by ITSP:  19998887777

pjsip.conf Configuration

We are assuming you have already read the Configuring res_pjsip page and have a basic understanding of Asterisk. For this NAT example, the important config options to note are local_net, external_media_address and external_signaling_address in the transport type context. The rest of the options may depend on your particular configuration, phone model, network settings, ITSP, etc. The key is to make sure you have those three options set appropriately.

local_net

This is the IP network that we want to consider our local network. For communication to addresses within this range, we won't apply any NAT-related settings, such as the external* options below.

external_media_address

This is the external IP address to use in RTP handling. When a request or response is sent out from Asterisk, if the destination of the message is outside the IP network defined in the option 'local_net', and the media address in the SDP is within the localnet network, then the media address in the SDP will be rewritten to the value defined for 'external_media_address'.

external_signaling_address

This is much like the external_media_address setting, but for SIP signaling instead of RTP media. The two external* options mentioned here should be set to the same address unless you separate your signaling and media to different addresses or servers.

Together these options make sure the far end knows where to send back SIP and RTP packets. This is important, because our Asterisk system has a private IP address that the ITSP cannot route to. We want to make sure the SIP and RTP traffic comes back to the WAN/Public internet address of our router. The contexts prefixed with "sipus" are all configuration needed for inbound and outbound connectivity of the SIP trunk, and the contexts named 6001 are all for the VOIP phone.