[asterisk-users] Re: NAT: RTP Path Optimization

[Conrad Wood]

On Wed, 2007-01-31 at 08:42 -0500, Andrew Kohlsmith wrote:
> On Wednesday 31 January 2007 8:28 am, Conrad Wood wrote:
> > This assumes all sip phones are set to reinvite=yes.
> > I expect (one of) the options to dial (tTw or W) to force asterisk to
> > remain in the media path. This way *only* if it's int<->int or ext<->ext
> > will it send sip reinvite, right?
> 
> Yes, but now you have to be careful of unintended consequences when people are 
> trying to use IVRs.

I wouldn't take it live as is without further testing, but I guess the
idea was worth adding to the thread.

> 
> What's wrong with having two peers, one with canreinvite=no, and Dial() using 
> the appropriate one?  (I haven't been following the thread, so this may have 
> already been discussed, and discounted.)
> 

That was the other idea I had - but think different Dial parameters are 
less problematic. For 2 peers per phone, the phones either need to have
a static public IP or need to be able to register with 2 credentials
(e.g. snom/cisco).


Conrad