Re: [LUG] Well that was ... Intersting (SIP conference call)

[Gordon Henderson <gordon+dcglug@xxxxxxxxxx>]

On Sat, 16 Jan 2010, tom wrote:

> Gordon Henderson wrote:
> > On Sat, 16 Jan 2010, Dan Dart wrote:
> >
> > > > Try having a small one before having a massive one... It will become
> > > > unmanagable above 4-5 people unless you are very, very disciplined.
> >
> > > True - 3 was bad enough but one was in another continent.
> >
> > That's where VoIP has a slight disadvantage over the PSTN - lag/latency.

> Not if its done 'properly'.

Well...

> The current tendency is to go through a hub or 
> exchange but theoretically once the call has been setup it should be able to 
> fallback to an ip to ip conversation  so if you were to talk to someone on 
> the same network 'branch' then the call wouldn't even have to go as far as 
> the exchange. Most stuff seems to be pretty primitive imitations of the phone 
> network - ultimately designed to get some money out of you somehow.

You've forgotten 2 things. One is NAT, (and there are ways to get round it 
- not always successful and less so when both ends are behind NAT which is 
going to be the case 99% of the time here) the other is that this is a 
conference bridge - the VoIP *has* to go via the bridge application/device 
to provide the one-to-many scenario that a conference entails.

(There are other issues where the VoIP needs to go via a centralised 
server such as call recording, transcoding in-audio DTMF signalling, but 
let's not wory about these here)

And even with SIP re-invites (which is what you're refering to), VoIP 
still has a higher latency than the PSTN. You probably won't notice it 
when all people are in the same country (or better, on the same ISP), but 
get someone abroad in the conference and you certinaly will notice it.

Gordon

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