[asterisk-users] Which features are lost when canreinvite is turned on ?

[Gordon Henderson]

On Sat, 7 Jul 2007, Olivier wrote:

> Hi,
>
> My setup is :
> PSTN --------- ISTP Network ----------- Router ------------- Asterisk
> ---------- SIP Phones
>
> Phones are located in the same location.
> I'm thinking about installing new phones in other locations (small agency,
> home workers), registering those phones to the same Asterisk server.
>
> As every location has DSL access, I think I should have those phones
> directly exchanging RTP data with ITSP media gateway, without passing
> through Asterisk server, with canreinvite = yes option.
>
> Before, trying this, I'm wondering which features I would loose in the
> process ?

The ability to pass audio between the endpoints if they are behind NAT 
firewalls...

You might be able to get it work, but I wouldn't bet on it.

See:

http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
http://www.voip-info.org/wiki/index.php?page=Asterisk+Letting+SIP+clients+connect+directly
http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

> Will I keep the ability to :
> - record CDR,
> - listen to DTMF tones
> - ...
>
> What do you think ?

I think it's challenging when NAT is involved!

Gordon