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

[Michelle Dupuis]

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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