[Asterisk-Users] Somewhat on topic but not * specific..
alex at opsys.com
Wed Mar 17 14:01:24 MST 2004
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
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?
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