From someone using Asterisk:
The m0n0wall Traffic Shaper (as far as I can grok it) by Adam Nellemann
On 9/26/05, Kristian Kielhofner <kris at krisk dot org> wrote:
> David Farrior wrote:
> > I have been trying to figure out how to do traffic shaping with my
> > asterisk@home setup. So far, I'm not having much luck.
> > After searching through the lists, it seems that the traffic shaping
> > wizard won't be much help. The rest of the m0n0wall documentation
> > is so good, iis there a good article out there that can thoroughly
> > explain this much needed function? If one already exists, please show
> > me where.
> > Your help is much appreciated.
> You might have some problems with Asterisk@Home. If you can edit
> sip.conf, iax.conf, etc. look for the tos= line. You should be able to
> set this to a fairly unique value. Actually, setting tos=lowdelay is
> the same as tos=0x10. You can then use the m0n0 traffic shaper config
> to match packets with TOS bits of "lowdelay". This COULD cause some
> problems because interactive SSH traffic (among other things) already
> sets the TOS bits on traffic to 0x10.
> It is by no means foolproof, but it's better than the alternative,
> which is to try to match traffic based on source and destination port
> numbers. With IAX this isn't too hard, but with SIP it is much harder.
> SIP is a call signaling protocol. RTP is the protocol used to
> actually transmit voice, video, etc. in conjunction with SIP (and other
> signaling protocols as well, less IAX). In Asterisk, the default is to
> use random UDP port numbers between 10000 and 20000 for RTP traffic.
> While this should be reduced, there is still nothing stopping other
> applications from using ports in this range and confusing m0n0,
> Asterisk, etc. I'm not sure how Asterisk@Home handles this, because I
> have never used it.
> The best solution would be to make a traffic shaper rule that ANDS the
> rules to combine TOS bits, source/destination IP addresses and port
> numbers (which m0n0 appears to be able to do).
> I'm glad you asked, because one of my pet peeves is when people think
> they are doing QoS with VoIP because they "prioritized" UDP port 5060
> (SIP signaling port), which actually does not do them any good
> (especially when using SIP + NAT).
> Sorry that this isn't very m0n0 specific - I'm new to m0n0 but old
> school Asterisk...
> Kristian Kielhofner
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