SIP NAT Routing issues - Asterisk 1.2 or Yate or SER?

Since everyone these days has firewalls (and frequently low budget ones) and NAT is virtually always in use, what is the best/easiest recommendation to get SIP extensions behind one fireall to properly register/stream with Asterisk across the net behind another firewall (the “worst case scenario” that is most frequent sceanrio)?

  1. Can/should Yate be setup as a dedicated SIP proxy between extensions and Asterisk?

  2. Will Asterisk 1.2’s rewritten SIP support finally finagle this frustrating SIP issue?

  3. Is there an easy How to to get SER to do it?

Have been trying to figure this out for a year, any help would be appreciated.



I’m pretty much a newbie at all this, especially the sip/nat stuff. But there are some pretty good threads going on over at voxilla (check this one, for example: … 0469#30469)
on this very subject. The user chandave is quite knowledgeable, and has been helping a few folks through various sip/nat gyrations.

Hope that helps.


The first thing i’d consider would be a tunnel or a VPN.

Tunnel/VPN is an expensive solution (even if easy/secure). Don’t really want to have to put VPN firewalls behind home wireless gateways & handle the IT as well as the costs, just to get an extension to light for infrequent calls.

Digium’s IAXy at $90 is a cheaper/easier solution - but forces analog handsets and has unadjustable loud/echo issues (we found out having just purchased one).