RE: Proposed Aim for Speak Freely

There's a lot to be said for sending 64k and
using an optimized larger bandwidth signal;
if optimized voice is the desired result.


-----Original Message-----
From: [email protected]
[mailto:[email protected]]On Behalf Of P Heyden
Sent: Saturday, August 16, 2003 10:14 AM
To: [email protected]
Subject: Re: [speak-freely] Proposed Aim for Speak Freely


Gentlemen,

I have followed this thread of what ought and what ought not be done with SF
and I only ask that the collective "we" not forget what many of us use SF
for.  Namely, "plain old internet telephone".

It is nice to talk about high quality codecs etc, but I can tel you that for
normal telecommunications 64K is not needed.  For many years I ran very good
quality PCM multiplexed voice circuits at 16K.  Our customers could not tell
the difference between 16k and 64K.

What is needed is a definitive plan for the future of SF.  As in many things
you have to walk before you run.  I see the maintenance and support of the
current product as a basic cornerstone for what the future might hold.

Sincerely,

Paul vander Heyden
----- Original Message -----
From: "Bob Arctor" <[email protected]>
To: <[email protected]>
Cc: <[email protected]>
Sent: Saturday, August 16, 2003 5:52 AM
Subject: Re: [speak-freely] Proposed Aim for Speak Freely


> yes , optionally.
> big drawback of SF is that you can't change quality/codec 'on the fly'
>
> i.e. over LAN where 10Mbit is a norm, and 10% is usually used bandwitch
isn't problem...
>
>
> since few months i am thinging about integration of Xvnc with speak
freely. but this would require high quality codeds in SF.
>
> Xvnc is remote desktop app, it would be great if it could transmit sound
also.
>
> i know there is NAS and ARTS/ESD but those servers do not support
compression+encryption.
>
>
> On Fri, 15 Aug 2003 10:11:16 +0200
> "Leonardo Boselli" <[email protected]> wrote:
>
> > If you want to trasmt music at a better quality it mis more likely that
you
> > want integrity than real-time . so a tcp trasmission would be bettere,
and
> > anyway this is not an internet phone application.
> > You can _optionally_ put other codecs, but for real time telephomy
> > saving lag and bandwidth it is a must.
> > Even if you had a wide bandwidth connection, usng a low bandwidth
> > channel would always yeld bettere real time results than larger
> > bandwidth ones (consider if in the same wideband connection there are
> > other users ....)
> >
> >
> >
> > Il 15 Aug 2003 alle 0:59, Bob Arctor immise in rete:
> >
> > > i disagree.
> > > when you talk over thing like inernet phone,
> > > your demands expand, perhaps you would like to send background
> > > noises, or you would like to have audio quality enough clean to be
> > > able to reproduce your voice when you move around the room ,
> > > or even wanted to transmit some music .
> > > there is ogg/vorbis compression, and mp3 compression, both which are
lossy. they introduce bit larger lag than usual GSM compression, but still
quite acceptable.
> > > they're meant for high-compression ratios, and sound compresed using
them sounds much better than using all other algorihms.
> > > you should check out mp3rtp.
> > > limiting bandwitch is good only in places where very small lag is
needed.
> > --
> > Leonardo Boselli
> > Nucleo Informatico e Telematico del Dipartimento Ingegneria Civile
> > Universita` di Firenze , V. S. Marta 3 - I-50139 Firenze
> > tel +39 0554796431 cell +39 3488605348 fax +39 055495333
> > http://www.dicea.unifi.it/~leo
> >
> >
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>
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