asterisk users - Feb 2004 - 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

Reading this thread leads me to chance asking a somewhat broad question 
of the gurus: is there a place in VoIP for multicasting?

Streaming scenarios, as well as conferencing, would seem to be ripe for 
that sort of integration, but I know nothing more about it beyond the 
fact that multicasting is oft-mentioned, but as far as I can tell (maybe 
kphone's vat hooks are an exception?) seldom implemented. . .

Thanks in advance for any light that might be shed.

B.

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@comcast.net>
To: <asterisk-users@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
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
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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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Marc Fargas
Sent: Monday, February 02, 2004 4:31 PM
To: asterisk-users@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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] En nombre de T. Chan
Enviado el: lunes, 02 de febrero de 2004 23:56
Para: asterisk-users@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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Adam Hart
Sent: Monday, February 02, 2004 5:28 PM
To: asterisk-users@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@comcast.net>
To: <asterisk-users@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
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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