[Asterisk-Users] Somewhat on topic but not * specific..

[Rich Adamson]

> On Wed, 2004-03-17 at 15:01, Alex Lopez wrote:
> > I have seen many postings today about the choppy sound problem. Some of
> > these problems were fixed with the recent change to rtp.c committed
> > today.
> > 
> > However in VoIP we usually do not have control of the quality of the
> > data pipe we travel over. I know there are tools that show sip proxies
> > traversed, how the IP packets reach to the desired endpoint.
> > (traceroute) but is there anything that can be used to 'rate' or
> > 'certify' that a route to a given endpoint has the bandwidth, speed,
> > lack of contention that would make for a good VoIP call?
> 
> Problems can be transient and therefore no tool would help unless it was
> receiving statistical packets from the remote side reporting on the
> quality of the packets you are sending.

The folks at www.netiq.com do have some tools that supposedly can help
model VoIP networks and diagnose problems, etc. I've not used any of them, 
but I've heard some other pbx manufacturers/resellers have been using
them. I'd have to assume they are used primarily on internal corporate
networks, and are probably not all that usefull for Internet-based
networks. 

I do use their Qcheck freebie, which was sold off to ixiacomm.com, and 
it is a rather handy tool for validating bandwidth between two points.
It relies on 'endpoint' software that can be remotely controlled from
a third point. I would guess NetIQ's voip tools are likely based on a 
combination of snmp and some additional endpoint software; snmp to 
interrogate the switches and routers between the endpoints, and the 
endpoints to measure jitter, delay and other voip parameters.

For those capable, it would be rather interesting to have some sort of
visual display of what * see's in terms of jitter; something like what
Brian through together to adjust x100p gain settings.

Rich