asterisk users - Nov 2006 - Direct UA to UA RTP connection

Greetings,

 

I have tried with all conceivable means to get my asterisk (called a in this
discussion) to have two SIP user agents (called ua1 and ua2 in this discussion
running SJPHONE actually) to communicate directly with one another using RTP. 
No matter what I do, the RTP traffic always goes between ua1 and a and a and
ua2, never ua1 to ua2 directly.  In my configuration a, ua1 and ua2 are all
within the same network with no NAT in between. Here are the asterisk
configuration settings I have:

 

Global

Nat=never (tried no also)

 

Sip peers

Nat=never (tried no also)

Canreinvite=yes

 

Once I get ua1 and ua2 to talk directly, I have another question.  If a, ua1 and
ua2 were all behind different NAT firewalls (ie a is in Boston, ua1 in Toronto
and ua2 in San Jose), what would it take to get ua1 to RTP traffic directly to
ua2.  In this last scenario, ua1 and ua2 are Linksys PAP2T devices.

 

Your expert help is greatly appreciated.

 

Mario

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