On Jan 16, 2009, at 10:38 AM, Gabriel Ortiz Lour wrote: > Hi all, > > Suposing that 2 SIP phone register at a remote (internet) > asterisk, what is the best way, if any, to make the RTP traffic go > phone to phone, whithout using the internet conection (asterisk)? Allow reinvite? Assuming both are not behind NAT.