[asterisk-users] Direct UA to UA RTP connection

[Mario François Jauvin]

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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