I think vonage is using g723.1 which requires 6.4kbps voice bandwidth compared to g711 - 64kbps.<br><br>For SIP to SIP calls, RTP doesn't necessarily goes thru the server. Only Signalling goes to the servers. This means no bandwidht usage for the provider.
<br>For SIP to PSTN calls, it has to goes thru a media gateway (owned by the provider) which may be seperate from the sip server. <br><br>Vikki.<br><div><span class="gmail_quote">On 11/2/06, <b class="gmail_sendername">Martin Joseph
</b> <<a href="mailto:ast@stillnewt.org">ast@stillnewt.org</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">On 2006-11-02 07:34:15 -0800, mail-lists <
<a href="mailto:mail-lists@peachnet.com">mail-lists@peachnet.com</a>> said:<br><snip><br>> My question is this: How do huge voip companies like vonage handle<br>> bandwidth. I'm pretty sure that they have to have sufficient bandwidth
<br>> available for X numbers of simultaneous calls, in other words ALL VOIP<br>> traffic runs through their servers, right? My boss is of the mind that<br>> there is no way that this is a viable business model and his insistence
<br>> has me doubting myself.<snip><br><br>For one thing, I suppose they use codecs that compress the voice data<br>as much as possible. Probably g729, or ilbc or some such.<br><br>Also, it's not true that all the traffic need to flow through there
<br>servers. Once the connections are setup in a well designed system, the<br>data could flow directly.<br><br>Marty<br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">
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