asterisk users - Mar 2004 - Somewhat on topic but not * specific..

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?

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.

Remember that the voice is going via UDP so it might get delayed for
other traffic, or even routed differently during the process of the
call. 
-- 
Steven Critchfield  <critch@basesys.com>