[Asterisk-Users] Can audio streams go client to cleint with IAX?

Adam Hart adam at teragen.com.au
Mon Feb 2 17:32:58 MST 2004 

what clients are you using? They probably won't support SIP

----- Original Message ----- 
From: "Marc Fargas" <asterisk at telenieko.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, February 03, 2004 11:18 AM
Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with
IAX?


> Thanks a lot I didn't see tose posts, so I'll have to migarte to SIP as I
> can see.. Do I need any other software or only Asterisk and my FXO's and
> FXS's firmwares migrated to SIP ?
>
> Thanks a lot.
>
> -----Mensaje original-----
> De: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] En nombre de Adam Hart
> Enviado el: martes, 03 de febrero de 2004 0:52
> Para: asterisk-users at lists.digium.com
> Asunto: Re: [Asterisk-Users] Can audio streams go client to cleint with
IAX?
>
> I think it would be fair to say Jeremy isn't really a fan of H.323 and,
> although he will fix bugs, (IMO) he won't be developing it anymore. You
> should give it a shot.
>
> ----- Original Message ----- 
> From: "T. Chan" <tommy.chan at utimail.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Tuesday, February 03, 2004 10:50 AM
> Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with
> IAX?
>
>
> > Would that be something that Jeremy would work on?
> >
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Adam Hart
> > Sent: Monday, February 02, 2004 6:36 PM
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with
> > IAX?
> >
> >
> > If you define possible as is H.323 capable of it, then yes.
> > If you define possible as is asterisk currently capable of it, then no.
> >
> > It wouldn't be too hard, look in channels/h323/ast_h323.cpp to see where
> > Jeremy started on it. You just have to get the openh323 lib to initiate
> the
> > transfer.
> >
> > ----- Original Message -----
> > From: "Marc Fargas" <asterisk at telenieko.com>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Tuesday, February 03, 2004 10:30 AM
> > Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with
> > IAX?
> >
> >
> > > Is it possible to make audio streams go client to client with H.323 ?
> > (both
> > > client being H323)
> > >
> > > Thanks!
> > >
> > > -----Mensaje original-----
> > > De: asterisk-users-admin at lists.digium.com
> > > [mailto:asterisk-users-admin at lists.digium.com] En nombre de T. Chan
> > > Enviado el: lunes, 02 de febrero de 2004 23:56
> > > Para: asterisk-users at lists.digium.com
> > > Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint
with
> > IAX?
> > >
> > > Dear All,
> > >
> > > Now, it seems that both IAX and SIP can have the two endpoints
> communicate
> > > the media directly without the media stream passing through the
> asterisk,
> > > can we do the same with H323 too?
> > >
> > > TC
> > >
> > > -----Original Message-----
> > > From: asterisk-users-admin at lists.digium.com
> > > [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Adam Hart
> > > Sent: Monday, February 02, 2004 5:28 PM
> > > To: asterisk-users at lists.digium.com
> > > Subject: Re: [Asterisk-Users] Can audio streams go client to cleint
with
> > > IAX?
> > >
> > >
> > > yes, IAX does direct transfers - when both ends confirm they can see
> each
> > > other, the asterisk server tells them to talk directly. With the
firefly
> > > network, we're seeing 90%+ connecting directly. Just to clarify, the
> audio
> > > doesn't separate from the call control.
> > >
> > > -Adam
> > >
> > > ----- Original Message -----
> > > From: "Jim Flagg" <flaggj at comcast.net>
> > > To: <asterisk-users at lists.digium.com>
> > > Sent: Tuesday, February 03, 2004 1:59 AM
> > > Subject: [Asterisk-Users] Can audio streams go client to cleint with
> IAX?
> > >
> > >
> > > > With a service like http://www.freshtel.net/?show=home that uses IAX
> and
> > > has servers in Australia,
> > > > is it possible for the  audio streams to take a different path than
> the
> > > call setup and control?
> > > > In other words can it work like SIP with canreinvite where the two
> > > endpoint negotiate audio
> > > > streams between themselves rather than though the FreshTel server?
> > > >
> > > > Thanks
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