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

Tue Feb 23 08:17:07 CST 2010

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Michelle Dupuis skrev:
> 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?

Unfortunately, that is not possible.

- - Tommy
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