[asterisk-users] directrtp with SIP + H.323

[wins mallow]

On Tue, 2010-02-23 at 08:22 -0500, Michelle Dupuis wrote:
> We're creating a SIP gateway for a client that will take one leg of a
> call in via SIP, and out the other side via H.323.  To minimize load
> on the gateway, we would like to have the RTP stream bypass the
> gatewayy altogether (directrtp/reinvite).  Is this possible with these
> to protocols?
>  
> Thanks
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IMHO, It's impossible ;) 

-- 
Best regards, Vince Mallow
xmpp: wins at jabber.slan.ru 
web: http://gentoo-way.blogspot.com