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

If you are migrating from chan_sip to chan_pjsip, then also read the NAT section in Migrating from chan_sip to res_pjsip for helpful tips.

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

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

Using RFC5737 documentation addresses

DeviceIP in example
VOIP Phone(6001)
ITSP SIP gateway

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 section and direct_media in the endpoint section. 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.


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.


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'.


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.


Determines whether media may flow directly between endpoints


Together these options make sure the far end knows where to send back SIP and RTP packets, and direct_media ensures Asterisk stays in the media path. 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 sections prefixed with "sipus" are all configuration needed for inbound and outbound connectivity of the SIP trunk, and the sections named 6001 are all for the VOIP phone.

For Remote Phones Behind NAT

In the above example we assumed the phone was on the same local network as Asterisk. Now, perhaps Asterisk is exposed on a public address, and instead your phones are remote and behind NAT, or maybe you have a double NAT scenario?

In these cases you will want to consider the below settings for the remote endpoints.


IP address used in SDP for media handling

At the time of SDP creation, the IP address defined here will be used as
the media address for individual streams in the SDP.
NOTE:  Be aware that the 'external_media_address' option, set in Transport
configuration, can also affect the final media address used in the SDP.


Enforce that RTP must be symmetric. Send RTP back to the same address/port we received it from.


Force RFC3581 compliant behavior even when no rport parameter exists. Basically always send SIP responses back to the same port we received SIP requests from.


Determines whether media may flow directly between endpoints.


Determine whether SIP requests will be sent to the source IP address and port, instead of the address provided by the endpoint.

Clients Supporting ICE,STUN,TURN

This is really relevant to media, so look to the section here for basic information on enabling this support and we'll add relevant examples later.

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  1. When your (remote) phone is behind NAT, you may want to check the UDP timeout in your gateway and adjust the "maximum_expiration" time in your phone's AOR settings, like this:


    If your router/gateway/modem is a Linux device with default settings, the UDP "stream" timeout default is 180, so 160 is a safe setting for your phone to re-register. If your UDP stream timeout is larger (/proc/sys/net/netfilter/nf_conntrack_udp_timeout_stream), you may adjust maximum_expiration accordingly.


  2. It should be noted that external_media_address and external_signaling_address currently do only allow for IPs as parameter until Asterisk 14.6 and 13.17.
    Once Asterisk 14.7 and 13.8 are released, this patch here should allow for dynamic hosts as parameter. Yay!


    1. A very useful info, thanks!