[Asterisk-Users] What is acceptable network latency for voip connection?

[Dorn Hetzel]

On Sat, Jan 08, 2005 at 11:01:25PM +0800, David Liu wrote:
> Well there is nothing much you can do if you don't own all the routes.  But in
> concept you can, and this is purely just theoritical and a very unhealthy
> thing for the Internet, is to write a program running on your router that
> constantly streams traffic to your end point, this will maintain a constant
> bandwidth from your network to your far-end.  Then, your program should detect
> within a few ms that you are setting a call up and immediately reduce your
> bogus traffic and make room for your "Real" voice traffic.  Again this is
> super unhealthy for the Internet, but the idea is TDM on STDM - constantly
> occupying certain trunks (bandwidth) on the Internet.  So whenever you need
> it, you will have it.  
> 
> David

David,

This is an almost unimaginably bad idea, and what's worse, it won't even
do what you want.  No matter how much data you stream constantly to "hold
your place", there is just no such feature present in the Internet.  Any
or all of the packets you send are always subject to being dropped due to
congestion.  Whatever happened one hour or one minute or one second or 
even one packet before is, for the most part (excluding routers with
route cacheing going on, and all that does is make your 2nd and later
packets do ever so slightly better than your first packet), completely
irrelevant.  Each packet gets forwarded or dropped at each router based
pretty much on the conditions that router sees right then;  Does it have
space on the outbound queue for the chosen interface, etc.  

So all this scheme will do is cause more lost packets in a global sense.

-Dorn