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