[Asterisk-Users] Can audio streams go client to cleint with IAX?
T. Chan
tommy.chan at utimail.com
Mon Feb 2 16:50:18 MST 2004
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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