as said, i've never used/needed the sunray audio input...

you may want to use:
===
$ /opt/SUNWut/bin/utsettings
$ sdtaudio
$ audiorecord x.au
$ gnome-sound-recorder
$ LD_PRELOAD=/usr/lib/libc_ut.so gnome-sound-recorder
===

gnome-sound-recorder may open /dev/audio directly (input on your physical ray 
server), which can be redirected to the correct (virtual) device by the last 
line (then it will open $UTAUDIODEV instead).

hope this helps,
niki

-------- Original-Nachricht --------
> Datum: Mon, 1 Jun 2009 10:55:18 +0800 (SGT)
> Von: fitra budi anggoro <[email protected]>
> An: SunRay-Users mailing list <[email protected]>
> Betreff: Re: [SunRay-Users] VOIP requirement - Upstream and downstream uttsc

> Hi Niki,
> Many thanks for the explanation. I can use your statement to counter my
> customer requirement :). 
> But to prove that our sun ray (dtu) can support audio input, any sw can I
> use on top of solaris 10 just to show them this working ?
> Thanks
> Fitra
> 
> 
> 
> 
> ________________________________
> From: Niki W. Waibel <[email protected]>
> To: SunRay-Users mailing list <[email protected]>
> Sent: Saturday, May 30, 2009 5:24:34 AM
> Subject: Re: [SunRay-Users] VOIP requirement - Upstream and downstream
> uttsc
> 
> hi fitra,
> 
> the actual problem here is, that RDP itself does not support audio input
> (audio transport from client to server) as of now. so neither (if anyone
> else knows better, please correct me) uttsc, nor rdesktop, nor mstsc client
> support audio input.
> 
> nevertheless, microsoft recently released all gory details about the RDP
> protocol:
> ===
> http://msdn.microsoft.com/en-us/library/default.aspx
> Open Specifications
> Windows Protocols
> Windows Communication Protocol ...
> ===
> and -- supprisingly -- there's a document titled
> ===
> [MS-RDPEAI]: Remote Desktop Protocol: Audio Input Redirection Virtual
> Channel Extension
> ===
> so it seems microsoft is working on this. i guess/hope, as soon as it is
> available on the server side, sun will implement this in uttsc.
> 
> the sunray protocol and the sunrays itself do support audio input since a
> long long time (2001?). i've actually never needed/tested it. this means if
> you use an audio application directly on linux and/or solaris, you should
> be able to use audio input.
> 
> hope the above explains some things.
> 
> rds, niki
> 
> -------- Original-Nachricht --------
> > Datum: Fri, 29 May 2009 10:44:37 +0800 (SGT)
> > Von: fitra budi anggoro <[email protected]>
> > An: SunRay-Users mailing list <[email protected]>
> > Betreff: Re: [SunRay-Users] VOIP requirement - Upstream and downstream
> uttsc
> 
> > Many thanks Aaron,
> > Looks like I am having a big problem here ;)
> > Currently I am installing SRSS on Solaris 10, and connect to Virtual Box
> > on Solaris 10 to host Windows XP host. None of your experience can be
> > implemented in my case :(
> > Anyway, again many thanks. I think Skip2Pbx or Vosky could be another
> > solution, but need to dig in more detail to syncrhonize with customer
> > rquirement.
> > Regards,
> > Fitra
> > 
> > 
> > ________________________________
> > From: Aaron Wilson <[email protected]>
> > To: SunRay-Users mailing list <[email protected]>
> > Sent: Friday, May 29, 2009 4:40:19 AM
> > Subject: Re: [SunRay-Users] VOIP requirement - Upstream and downstream
> > uttsc
> > 
> > Looks like no one else responded. I don't really have an answer. I can
> > just tell about my experiences with it.
> > 
> > We never were able to figure out how to use a sunray for any kind of
> > stable and usable VoIP in my office. I don't think there is a simple
> solution.
> > Only pure Skype program method I can see being feasible is run Linux and
> > SRSS. We never tested it though, but there is a Linux version of Skype.
> > 
> > We were able to get SIP Communicator on Solaris running in a testing
> > environment connected to an Asterisk box. Asterisk and Skype have now
> partenered
> > up. That program is still in private beta. Skype to Sip is another
> option
> > Skype is offering now in beta. I believe it just creates a sip trunk
> > between a SIP capable PBX and Skype.
> > 
> > In my office we opted for a solution that connected Skype to our
> existing
> > PBX. We didn't want our users to have to man a desk phone and a soft
> phone.
> > We also didn't want them to be able to IM with friends all day, which
> they
> > can do with the Skype client. Lastly we wanted our receptionist to be
> able
> > to tell when our employees are on call whether that be a Skype call or a
> > regular landline call. We have a No Microsoft policy so we opted for
> > skip2pbx versus other skype to pbx options out there. Skip2pbx is
> ubuntu/debian
> > based. We looked at Vosky too but their option was more than 4x the
> price of
> > skip2pbx. Of course Vosky is a turn key solution. skip2pbx is almost
> > identical in features, but does require you to provide your own hardware
> and
> > knowledge getting everything running.
> > 
> > Of course this solution isn't perfect when there is a desk with a sunray
> > but no desk phone....
> > 
> > And like I said we are Microsoft free, so Skype on Windows was never an
> > option so we never explored that path.
> > 
> > If you ran SRSS on Linux or OpenSolaris then the Ekiga SIP client would
> be
> > on the system. Not so on Solaris 10. I can only assume two way audio
> would
> > work with Ekiga as I've never tested it. Gizmo on Linux might be another
> > possibility. It too can now connect to the Skype network.
> > 
> > 
> > Aaron Wilson
> > Ernie Ball, Inc.
> > I.T. Department
> > 
> > 
> > fitra budi anggoro wrote:
> > > Dear All,
> > > Got a requirement from our customer to run VoIP in Windows (eg Skype) 
> > Hence the need for both upstream and downstream audio.  The RDP client,
> > uttsc, does only downstream (ie to the DTU) today.  Upstream from the
> DTU audio
> > in (to the server) is not there yet.
> > > Any idea how to meet this?
> > > Thanks
> > > Fitra

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