[asterisk-dev] RTP streams between H323 and SIP

[Paul Cadach]

Hello,

Johansson Olle E wrote:
> > Are you sure? All channels that supply 'ast_rtp_bridge' as their
> > bridge method should already support it.
>
> That was news to me. Cool if that's the case!
> Never seen or heard about it before.
>
> I stand corrected.

I just acknowledge we have H.323 native bridge code that support RTP "move" (like re-invites for SIP) and all works fine
between SIP and H.323 channels.


WBR,
Paul.