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

[Adam Tauno Williams]

>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.