46 lines
1.9 KiB
HTML
46 lines
1.9 KiB
HTML
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2 Final//EN">
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<HTML>
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<HEAD>
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<META NAME="GENERATOR" CONTENT="SGML-Tools 1.0.9">
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<TITLE>VoIP Howto: Bandwidth consideration</TITLE>
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<LINK HREF="VoIP-HOWTO-10.html" REL=next>
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<LINK HREF="VoIP-HOWTO-8.html" REL=previous>
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<LINK HREF="VoIP-HOWTO.html#toc9" REL=contents>
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</HEAD>
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<BODY>
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<A HREF="VoIP-HOWTO-10.html">Next</A>
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<A HREF="VoIP-HOWTO-8.html">Previous</A>
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<A HREF="VoIP-HOWTO.html#toc9">Contents</A>
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<HR>
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<H2><A NAME="s9">9. Bandwidth consideration</A></H2>
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<P>From all we said before we noticed that we still have not solved
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problems about bandwidth, how to create a real time streaming of
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data.
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<P>We know we couldn't find a solution unless we enable a right
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real-time manager protocol in each router we cross, so what do we
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can do?
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<P>First we try to use a very (as more as possible) high rate compression
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algorithms (like LPC10 which only consumes a 2.5 kbps bandwidth,
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about 313 bytes/s).
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<P>Then we starts classify our packets, in TOS field, with the most
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high priority level, so every router help us having urgently.
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<P>Important: all that is not sufficient to guarantee our conversation
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would always be ok, but without an great infrastructure managing
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shaping, bandwidth reservation and so on, it is not possible to do
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it, TCP/IP is not a real time protocol.
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<P>A possible solution could be starts with little WAN at guaranteed
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bandwidth and get larger step by step.
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<P>We finally have to notice a thing: also the so called guaranteed
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services like PSTN line could not manage all clients they have: for
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example a GSM call is not able to manage more that some hundred or
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some thousand of clients.
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<P>Anyway for a starting service, limited to few users, VoIP can
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be a valid alternative to classic PSTN service.
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<HR>
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<A HREF="VoIP-HOWTO-10.html">Next</A>
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<A HREF="VoIP-HOWTO-8.html">Previous</A>
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<A HREF="VoIP-HOWTO.html#toc9">Contents</A>
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</BODY>
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</HTML>
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