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 From:  sai <sonicsai at gmail dot com>
 To:  Kristian Kielhofner <kris at krisk dot org>
 Cc:  David Farrior <davidfarrior at gmail dot com>, m0n0wall at lists dot m0n0 dot ch
 Subject:  Re: Traffic Shaping - I'm Confused
 Date:  Mon, 26 Sep 2005 10:01:22 +0500
From someone using Asterisk:
http://www.mollien.net/index.php?main=articles&article_id=24

The m0n0wall Traffic Shaper (as far as I can grok it) by  Adam Nellemann
http://m0n0.ch/wall/list/?action=show_msg&actionargs%5B%5D=49&actionargs%5B%5D=99

sai



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.
>
> David,
>
> 	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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