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

[Jeremy Jones]

Generally speaking, unless you're using an rtp proxy, the rtp audio
should go client<-->client.  H323 does the call setup and teardown and
such, but the audio stream is usually direct.

Jeremy 

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Marc Fargas
Sent: Monday, February 02, 2004 4:31 PM
To: asterisk-users at lists.digium.com
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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