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

[David Liu]

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



On Sat, 8 Jan 2005 06:22:58 -0800 (PST), Robert Augustyn wrote
> Very good point.
> So what can you do ( if anything ) to control the load
> on the network outside of your control?
> robert
> 
> --- David Liu <david at deltapath.com> wrote:
> 
> > Assuming the network loading is fairly constant,
> > 300ms latency is actually not
> > noticeable unless you put both phones next to your
> > ears to compare.  
> > 
> > Latency affects delay while network loading affects
> > voice quality (e.g. break
> > ups) If the either end of your network is
> > experiencing very bursty traffic
> > patterns, then even a small latency won't
> > necessarily guarrantee good sound
> > quality.  
> > 
> > David Liu
> > Hong Kong
> >