Re: [GnomeMeeting-devel-list] H.350 server
- From: Alexandre Aractingi <aaractingi libertysurf fr>
- To: GnomeMeeting development mailing list <gnomemeeting-devel-list gnome org>
- Subject: Re: [GnomeMeeting-devel-list] H.350 server
- Date: 11 May 2004 14:15:38 +0200
Le mar 11/05/2004 à 14:08, Damien Sandras a écrit :
> Indeed but it doesn't solve the above problem. Well sort of, by having
> all H.323 users registering to a GK, all SIP users registering to a SIP
> registrar, Asterisk could bridge the calls and permit to H.323 users to
> call SIP users even if SIP is disabled on the H.323 side and vice-versa.
Just a sidenote on the way Asterisk deals with H.323: as far as I know,
the current H.323 interfaces in Asterisk (chan_h323 and asterisk-oh323)
both force the media (RTP) to go through Asterisk once the call is
established. They don't just route the signalling channels, so this
solution might quickly overwhelm a publicly available Asterisk server at
Seconix.
A bit OT, but I thought it was worth saying it :-)
--
Alexandre Aractingi <aaractingi libertysurf fr>
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