[Asterisk-Users] Digium should develop and sell just Dummy card. For timing...

[Chris Albertson]

Lets see if I understand this logic.  I'll restate it:

1) Asterisk's MOH is only broken if you attempt to build a
   VOIP-only system
2) Asterisk is not intended for such use. It is a PSTN
   oriented PBX that just happens to handle VOIP.
3) Therefore Asterisk is not broken

OK. If you believe #2 you are right.

My comment was outside of the above argument.  Let me restate
my argument that is intended to counter those thinking timing
hardware should be required

1) There is no requirement that outbound VOIP media packets be
   precisly timmed
2) Therefore there is no reson to use a hardware timmer card.
3) Software that depends on a hardware timmer card
   for timming outbound VOIP media is doing so needlessly.

To support the above argument you can look at the RFCs to support
#1 and you can look at other VOIP software to support #3

The bottom line is that it would be silly for anyone to offer a
timming-only PCI card

--- Steven Critchfield <critch at basesys.com> wrote:
> On Wed, 2003-10-15 at 12:50, Chris Albertson wrote:
> > 
> > Steven,
> > 
> > Good comments but remember good enginerring starts with reading
> > the requirements and desiging to those requirements.
> > in the case of SIP at least these is an RFC.  What is the
> > timming requirement on media packets?  How is the stream
> > synchronized?  I'll read it in the next few days but I'd
> > bet a beer there is no requirement for precise timming.
> > 
> > Any discusion about PCI cards, RTC timmers and the like is in
> > a complete vacuum unless you know what exactly it is that the
> > software is required to do.
> 
> And this means what to a system designed around a PSTN board? How
> many
> of us use this system for PSTN and IAX with no SIP or h323 in sight? 
> 
> So far these timing problems arise from people doing VoIP only
> systems
> on a system that was started as a PSTN platform and is able to handle
> VoIP.
> 
> So while reading the RFC is a good idea, and maybe it will be good in
> determining a new timing source for SIP only systems, it shouldn't
> take
> over asterisk.
> -- 
> Steven Critchfield  <critch at basesys.com>
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users


=====
Chris Albertson
  Home:   310-376-1029  chrisalbertson90278 at yahoo.com
  Cell:   310-990-7550
  Office: 310-336-5189  Christopher.J.Albertson at aero.org
  KG6OMK

__________________________________
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search
http://shopping.yahoo.com