[Asterisk-Users] sip to h323 ... direct RTP?

[Cesc]

On 6/21/06, Kevin P. Fleming <kpfleming at digium.com> wrote:
>
> ----- Johansson Olle E <olle at voop.com> wrote:
> > No. It's certainly possible but at this time there's no interaction
> > between
> > the RTP clients, the various channel drivers.
>
> I believe this is incorrect; all the RTP-using channel drivers supply 'ast_rtp_bridge' as their native bridge method, so assuming they also implement the 'set_rtp_peer' method, then an RTP native bridge between dissimilar channels should work fine. If the channel driver(s) also support sending the RTP peer address to the endpoint (as chan_sip does with reinvite), then a direct media path should also be possible.
>

Sorry, but I am confused.
What does this mean for the bare-bones user like me? That technically
it would be possible but that it is not implemented? (i use the sip
and h323 channels shipped with the latest sources tarball) Or that it
is possible to configure via some obscure setup file?

Thanks!

Cesc

> --
> Kevin P. Fleming
> Senior Software Engineer
> Digium, Inc.
>
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