Inbound QoS with virtual ethernet and policy routing

2017 December 11
by Daniel Lakeland

Starting in mid october I began working on many projects involving computer networking. When you have a fairly complicated set of computing services you just need to do maintenance on them, and also inevitably there's something wrong with some of them that needs fixing, but it takes concentrated time, so you put it off...

Anyway, I made a bunch of things better, including my SIP based VOIP system, and my internal computer network in my house which is now ipv6 only except for certain legacy devices. Ipv6 solves much of the problems that SIP has with NAT, and so this was an overall win. In the process of all of it, I spent some time updating my routers, and got involved with the LEDE project (a fork of OpenWRT). In the process of that I started helping someone on the forum understand QoS, and learned something about setting up QoS myself.

The problem:

Inbound QoS is normally considered "impossible" because once you have a packet you can't keep it from being sent to you... and so the best thing to do is just forward it as quickly as possible. This is not true. In fact it relies on a mistaken idea of what QoS is all about. QoS is about increasing the "utility" of your network by selectively slowing down things that you care less about so that things that you care more about go faster than they would if you didn't slow down those unimportant things.

The "value" of a stream of packets is related to how important they are to you, not how many of them you get per second. If a download you plan to run all night while you sleep is slowed down so your voice conversation is crystal clear, even though you're passing fewer packets over the wire per second, your quality of service increases.

Inbound QoS in Linux:

In Linux, the receive queue has far fewer features than the send queue on an interface. Also, the receive queue has to make decisions before Linux has sent the packet through iptables and had a chance to use marking or DSCP setting iptables rules (such as in the mangle/PREROUTING table). The logical thing to do with inbound QoS is to put it in the router that routes to the wide internet, and run the packets through the iptables and then put the packets into the LAN output queue with appropriate tags and all the capabilities of the output queue. The problem comes when you have a router with several interfaces. For example maybe it has two separate LANs or it has 2 wifi interfaces and an ethernet interface all bridged together. You can't control the total bandwidth by setting limits on the individual interfaces. You want to control the total bandwidth though, because this is what comes in your WAN pipe.

Solution:

Suppose you have a bridged LAN with 3 physical interfaces, such as you might on a LEDE/OpenWRT wifi router. Let's call the interfaces wlan0, wlan1, and eth0 and they're bridged into a bridge called br-lan. Let's suppose there's also an interface "wan" where packets come in from the internet.

We can force packets received on wan to be routed to a virtual ethernet pair veth0 and veth1. This is based on where they're received from, and so is part of Linux policy routing. The way a veth works is that anything sent to one of them immediately is received by the other as if they were a pair of ports on your computer with an ethernet patch cable between them. To set this up we can do something like:

ip link add type veth ## by default creates veth0 and veth1
ip link set veth1 promisc on ## might be unnecessary
ip link set veth1 master br-lan ## bridge veth1 into the lan

ip route add default dev veth0 table 100 ## send everything that uses table 100 to veth0
ip rule add iif wan table 100 priority 100 ## if it comes in wan interface use table 100 for routing

Now packets that come in wan go through the iptables where we can have -j DSCP rules that tag them with DSCP values describing their importance, then the packets hit the routing code, the code looks up the rule priority 100 and sees it applies so it uses table 100 to route the packets, and table 100 says to send everything down veth0 towards veth1. Since the packets have been through the iptables before routing, by the time they hit veth0 we can for example use fireqos to set up the output of veth0 to have queues which shape the traffic, in particular they delay and/or drop the traffic we don't care about as much which has less important DSCP tags.

There are some tricks here though. The bridge itself will send packets through the iptables again if we have certain sysctls enabled. This may filter the traffic so that it doesn't go from veth1 into the bridge. If you don't want that you need in /etc/sysctl.conf

net.bridge.bridge-nf-call-arptables = 0
net.bridge.bridge-nf-call-ip6tables = 0
net.bridge.bridge-nf-call-iptables = 0

Or if you do want it, you need to set up your iptables to allow the appropriate traffic.

The delay/dropping of packets that are low priority is critical because we may not be able to stop people from sending us packets we've already received, but because of TCP window and the ACK feedback, a short time in the future, if we haven't sent back ACK packets, they will slow down their sending rate. So with TCP in the mix, shaping your inbound packet stream results in feedback going back to the sender in the form of delays in acknowledgement and then slowdowns on their send rate. Explicit congestion notification can also help here. So it's not good to just forward your packet as fast as possible on inbound... you want to give that feedback in the form of delayed ACKs to tell your other party to slow down and open up the channel because you're using some of it for something else.

The result is maybe the very beginning of your voice call might have a little garbled audio, but after say 1 second the delays on your download caused at the veth0 output queue result in your download partner throttling the download, and then your audio packets can come in and will be fine for the rest of the call. Similarly for latency sensitive games, providing you're tagging appropriately.

Tagging your high priority traffic with DSCP tags as it comes in your router is also a good idea because downstream inside your LAN you may have managed switches or wifi access points which have WMM queues and they will interpret the higher priority DSCP tags to mean they should provide basic QoS as well. (note all 802.11n or later, including 802.11ac access points use WMM for prioritizing different traffic, this also changes the radio-collision algorithms used)

Because of the way linux wifi drivers work, it's a good idea to use DSCP tag CS6 = decimal 48 on your highest priority traffic, such as voice RTP traffic. This isn't the standard, the standard for voice seems to be EF = decimal 46 but this doesn't in general cause Linux drivers to use the WMM Voice queue. So setting up DSCP 48 on your softphones and retagging to 48 as packets come in from the internet is probably best.

Hopefully this helps some people. I'll be linking this post to FireQOS issues so people can find it that way.

 

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