Update: 02/2015 - This is a legacy article specifically for PSTN Gateway (Trunking) services, not Hosted PBX. For Hosted PBX information regarding NAT, please see the Router Configuration Section and choose your specific router.
The traversal of NAT (Network Address Translation as defined in RFC 1631) devices is a major problem for the widespread deployment of VOIP; SIP based VOIP in particular. The issue is non-trivial and there are no simple solutions.
In general terms there are two ways to deal with this problem:
- avoiding the problem altogether
- working around the problem
The best way to deal with the issue of VOIP NAT Traversal is to avoid the cause of the problem in the first place; that is, simply do not use a NAT. An alternative to avoiding NAT is to avoid SIP; that is, you could use a NAT friendly protocol such as IAX.
If you cannot avoid the problem, successful communication can be achieved in most cases by utilizing one or more of several techniques available to work around the problem. OnSIP recommends trying the following:
- Use SIP keep-alive packets or set your Registration interval below the NAT binding expiration time (90 seconds will usually suffice).
- Set the "Symmetric RTP" option to "Yes" in the SIP Configuration section of OnSIP User Portal (assumes Symmetric RTP is being utilized).
Additional techniques such as port forwarding, STUN, and manual IP address configuration are available. The following white paper explains some of the issues and workarounds in more detail: NAT Traversal in SIP