[Asterisk-Users] QoS in asterisk

[steve at daviesfam.org]

On Tue, 13 Jul 2004 matt.riddell at sineapps.com wrote:

> Qualify will only stop the call going through if for example the ping 
> is above 200ms.  I find most of my problems come from fluctuating 
> ping times (~100ms) than from a stable high ping.  

I agree that the overall delay isn't really the problem - jitter and 
packet loss are what causes the trouble.

There really isn't currently anything in Asterisk which measures this - 
especially not when there is no active call using the path.

The IAX2 jitter buffer code does know the amount of jitter - and could
probably make this measurement available in a variable or something. And I
propose to add similar jitter buffer code for SIP and other RTP-using
protocols too.

But I'm not really sure how the measurement can then be used effectively
for call routing.  I'd be interested in your ideas.

Note that I observe that in my environment jitter and packet loss come and 
go over a timescale of seconds - this a result of sharing a narrowish pipe 
with a bunch of other traffic without any shaping to help the VOIP 
traffic.

For this environment the real fix is to improve the network rather than do 
anything too complicated with *.  (Not to say that *s jitter handling and 
packet-loss-concealment can't be improved - I've been working on that and 
I'm still busy).

I'm about to ask for some help in gathering jitter stats from a bunch of 
users - perhaps you'd like to help with that.

Steve