[Asterisk-Users] P2P RTP without SIP re-invites

[Brancaleoni Matteo]

Hi.

> If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes
> * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it
>    goes directly from Alice to Bob
> Not all UAs support a re-INVITE and in public scenarios, a lot of UAs have broken NAT support so
> the RTP media stream stays with Asterisk.

a small correction: doesn't matter if Alice and Bob are nat'ed:
if they're both nat'ed re-INVITEs are sent and RTP is transferred
to go directly from Alice to Bob. Asterisk manages only the
signalling on port 5060
(I'm using that environment, so it works :) )

But if only Alice OR Bob are nat'ed, the RTP is handled by * itself.

Matteo.

-- 
Brancaleoni Matteo <mbrancaleoni at espia.it>
Espia - Emmegi Srl