[asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination

[Terry Wilson]

> The setup is as follows: SIP phone registers via international link  
> to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2  
> via Zaptel Channels need to be hairpinned from Box 1 to 2. How is  
> sip.conf configured on Box 1 and 2 so that we don't get an error:  
> "Failed to authenticate user" when 1's extensions.conf uses SIP to  
> dial Asterisk Box 2 . How do we ensure that RTP traffic flows from  
> SIP phone registering at 1 directly to 2 without first passing  
> through 2?

I think if you set up a peer for Box 1 on Box 2, and set insecure=port  
on those peers, that it will not try to auth calls that are from your  
other asterisk box.  Of course, you'd have to make sure in your  
diaplan that you restricted access to those calls appropriately.  For  
the RTP, setting canreinvite=yes one peers that you want to be able to  
send media directly to each other should allow the RTP behavior you  
are looking for, but keep in mind that if there are any NATs between  
the phones, things can get messy in a hurry.