"Stephane M" <Stephane@M.com> wrote in message
news:461A9DBF.6020302@M.com...
> Thanks guys for all this information...
>
> But I think about companies, which want to install a VoIP system....
> that seems to me not really suitable for companies..

you are assuming that the VoIP traffic is for everything - there are lots of
variations.

there are IP Telephony / Voip systems out there that only use IP across a
LAN, and / or use a corporate WAN where QoS can be imposed across all
traffic with a central voice gateway.

And some telcos (not necessarily ISPs) will provide voice "trunks" via IP -
sort of gateway in the carrier core network.

>
> you can't ask people to stop downloading while phoning :-)

No - but you can design the combined network so it doesnt matter.
>
> So, I can't really see any possibility for professional use... Seems to
> be a bit dodgy !??

i have worked on IPT / voip systems for big call centres with several 100
users

it works really well, and is more flexible in some ways than a conventional
phone system (or they wouldnt have put it in....)
>
> I installed in the company an FVX538, thinking that I would be able to
> use VoIP systems...
>
> So, the only solution would be the CTX1000 ???

the issue is not what kit you use, but that incoming voice packets didnt get
"high priority / low delay / low jitter" treatment on their way to you.

Nothing you do to outbound traffic can completely solve that, although a few
of the workarounds suggested might improve your case.

Even then if something out there is flooding packets at your router, nothing
you do will turn off the inbound traffic, so any voice or other traffic you
do want to come in will struggle.
>
> Stephane
>
> Gordon Henderson a écrit :
> > In article <evdfic$24c$1$8300dec7@news.demon.co.uk>,
> > Stephane M <Stephane@M.com> wrote:
> >> Hi Tim,
> >>
> >> Thanks for your feedback
> >>
> >> I am bit surprised that the QoS is not so good actually...
> >> Anyway...
> >>
> >> I thought that SIP was the protocol for all VoIP communications ? I
> >> mean, I thought that even the the Audio of the communciation, used this
> >> protocol... !?!
> >
> > SIP is the Session Initiation Protocol. It controls the signaling
between
> > the end-points. It does authentication, sends the dial codes, and lets
each
> > end tell each other (to some extent) how to carry the call data between
> > each end-point.
> >
> >> Anyway, I found on ny phone, that RTP port is [ 16384 - 16482 ]
> >> So I added, on my QoS service..
> >>
> >> 1) Do you know which protocol is using RTP ? (TCP/IP ???)
> >
> > RTP is the protocol. (Realtime Transport Protocol) It's UDP.
> >
> > One issue you'll never be able to solve, no matter what router you use
> > is that once your data is out on the Internet you have absolutely zero
> > control over it.
> >
> > All you can effectively do at your end is control data leaving your
> > network. If you do large uploads, (data leaving your network) then you
> > can prioritise your VoIP traffic over this.
> >
> > However one thing you can't do is prioritise incoming VoIP traffic over
> > any other incoming traffic, and the reason for this is simple - by the
> > time you get the packet and decide to delay it (if it's not a VoIP
packet)
> > it's too late. That packet has already come over the wire. There are
> > tricks that can help, but you can never successfully prioritise incoming
> > traffic, especially stuff with real-time contraints.
> >
> > So if you are having issues, the best thing to do is simply shut down
> > all traffic when making calls. Not much help when a call comes in
though,
> > and once on the 'net, it's anyones guess what'll happen to the packets,
> > so make sure you use a decent ISP who doesn't have a conjested internal
> > network of their own, and be prepared to pay for it!
> >
> > Gordon
--
Regards

stephen_hope@xyzworld.com - replace xyz with ntl