From all we said before we noticed that we still have not solved
problems about bandwidth, how to create a real time streaming of data.
know we couldn't find a solution unless we enable a right real-time manager
protocol in each router we cross, so what do we can do?
we try to use a very (as more as possible) high rate compression algorithms
(like LPC10 which only consumes a 2.5 kbps bandwidth, about 313 bytes/s).
we starts classify our packets, in TOS field, with the most high priority
level, so every router help us having urgently.
all that is not sufficient to guarantee our conversation would always
be ok, but without an great infrastructure managing shaping, bandwidth
reservation and so on, it is not possible to do it, TCP/IP is not a real
solution could be starts with little WAN at guaranteed bandwidth and get
larger step by step.
finally have to notice a thing: also the so called guaranteed services
like PSTN line could not manage all clients they have: for example a GSM
call is not able to manage more that some hundred or some thousand of
for a starting service, limited to few users, VoIP can be a valid alternative
to classic PSTN service.
this content on VoIP was taken from VoIP Howto, written by Roberto Arcomano
firstname.lastname@example.org on v1.7, August 7, 2002. The most up to date version
of this information can be found at: http://www.fatamorgana.com/bertolinux
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