Sip + nat

Hello I`m using asterisk 1.4 with 2 sipura clients. My asterisk has a public IP address, and my clients are behind a router wich makes NAT. In sip.conf I set nat=yes and canreinvite=no for both users. I thought they wasn´t going to work without a stun server or somethink like that, but I could make a call between the users!. Can someone explain that? or tell me a link wich explain that?

I tried to do that in asterisk 1.2 and it didn´t work without a stun server for my sip clients! What change in asterisk 1.4 related with nat for sip users??

Thanks in advance