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

[Kristian Kielhofner]

On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis <support at ocg.ca> 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

Yate claims it can do this:

http://yate.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy

-- 
Kristian Kielhofner
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