[asterisk-users] Re: VOIP Bandwidth questions

[Vikki]

I think vonage is using g723.1 which requires 6.4kbps voice bandwidth
compared to g711 - 64kbps.

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.
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.

Vikki.
On 11/2/06, Martin Joseph <ast at stillnewt.org> wrote:
>
> On 2006-11-02 07:34:15 -0800, mail-lists <mail-lists at peachnet.com> said:
> <snip>
> > My question is this: How do huge voip companies like vonage handle
> > bandwidth. I'm pretty sure that they have to have sufficient bandwidth
> > available for X numbers of simultaneous calls, in other words ALL VOIP
> > traffic runs through their servers, right? My boss is of the mind that
> > there is no way that this is a viable business model and his insistence
> > has me doubting myself.<snip>
>
> For one thing, I suppose they use codecs that compress the voice data
> as much as possible.  Probably g729, or ilbc or some such.
>
> Also,  it's not true that all the traffic need to flow through there
> servers.  Once the connections are setup in a well designed system, the
> data could flow directly.
>
> Marty
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061102/8d1d88bc/attachment.htm