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

[Adam Hart]

Sure, if you want to code it :) Wouldn't be that hard either. Actually what
would be really sexy (if that's the word) is native transfers between H.323
and SIP given they both use the same RTP (correct me if I'm wrong on that)

----- Original Message ----- 
From: "T. Chan" <tommy.chan at utimail.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, February 03, 2004 9:56 AM
Subject: 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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