[Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?

[wasim at convergence.com.pk]

before critch looses his top at the redhat links below... the originating
source for these documents is at http://lartc.org

- wasim

On Tue, 13 May 2003, Adam Tauno Williams wrote:

> >The short answer is "VOIP RTP is UDP, which normally pushes TCP 
> >sessions out of the way, so most of the time everything is OK."
> 
> With respect, I could hardly disagree more.  We have a network of frame relay,
> point-to-point T1's and ISDN circuits, and without QOS VOIP service was spotty
> at best.  Our network provides adequate (and cost effective) bandwidth for data,
> but voice quality was tortuous.  With the addition of QOS voice quality is
> nearly pin-drop quality without data rates bieng significantly effected.
> 
> > The long answer is that you need to look at QOS (Quality Of Service) 
> > and how your particular router vendor implements it.  On the "global" 
> > Internet, you're pretty much out of luck since very few providers 
> > exchange QOS information to ensure end-to-end priority for packets 
> > with TOS (Type Of Service) bits set in their headers.  If this is all 
> > within your own control, you should talk to your router vendor (or 
> > your router technician) and see what they can tell you about how to 
> > implement QOS across your network.
> 
> If your voice traffic uses a specific range of ports (usually pretty easily
> accomplished with most products) than adding priority queuing at router devices
> based upon port number should be the simplest solution.
> 
> See -
> http://www.redhat.com/mirrors/LDP/HOWTO/Adv-Routing-HOWTO/lartc.qdisc.filters.html
> and you should be able to whip something together.
> 
> It really is required to implement this on the routers that bear the traffic,
> the host doesn't really have all that much to do with it - although there are a
> couple of ways to 'mark' packets for high priority delivery.  But port based
> prioritizing has proven more than sufficient for us.
> 
> > Realistically, it's only on congested networks that QOS is 
> > meaningful, anyway, unless you're doing some freaky 
> > least-cost-routing tricks with your transit providers (at any layer). 
> > If your network is at <70% capacity during peak minutes (it is, 
> > right?) then probably QOS is not going to be necessary.  To give you 
> > an idea: I regularly use a VPN over a cable modem to connect to a SIP 
> > gateway 3500 miles and 130ms away, with zero voice artifacts and no 
> > noticeable quality loss, and this is ALL over the public Internet, 
> > with no QOS implemented.
> 
> Agree, voice of the Internet works pretty well minus QOS.  Voice over something
> like a frame relay net *REQUIRES* QOS, as data rates are pretty good but
> fluctaute almost instantaneously, and the only thing you can count on is your
> commited rate.
> 
> >>Just wondering if anyone out there has done any work, or knows where 
> >>any work is being done, to try to honor the latency requirements of 
> >>this VOIP stuff and push out SIP and RTP traffic, etc., "ahead of 
> >>the crowd."
> 
> http://www.redhat.com/mirrors/LDP/HOWTO/Adv-Routing-HOWTO/index.html
> should tell you everything you need to know.
> 
> >>I'm doing my VOIP behind wireless, so it is particularly important. 
> >>I am getting ready to do some digging, and don't want to re-invent 
> >>the wheel.
> 
> If you wireless isn't congested and the signal is consistent it is more than
> enough bandwidth.  But a misbehaving client can ruit it for you, and there is
> little to nothing you could do about it.  You can prioritize out the the
> wireless from you "backbone" but incoming traffic is just going to be hit-n-miss.
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