MikroTik

Hello,

I have been working with a Mikrotik technician to configure our RouterBOARD 750G's to be used in camps that have data and phones going over a satellite link. The links are @ 1Mbps down and 200kbps up.

I am stress testing the voip using loic.exe as well as flooding the network with large ICMP packets. So far whatever I try the voice quality degrades to the point where the called party cannot hear me at all. Ideally, the Mikrotik RouterBOARD 750G should see the voip-control-down, voip-rtp-down, voip-control-up and voip-rtp-up as having a higher priority as the un-tagged data, forward the voip and buffer and or drop the data packets. Obviously that isn't the case.

Here is the sample of /queue tree> print

0 name="VoIP-control-Down" parent=Phone packet-mark=voip-control limit-at=1>
queue=default priority=3 max-limit=2M burst-limit=0 burst-threshold=0
burst-time=1s

1 name="VoIP-rtp-Down" parent=Phone packet-mark=voip-rtp limit-at=1M
queue=default priority=1 max-limit=2M burst-limit=0 burst-threshold=0
burst-time=1s

2 name="Network-Down" parent=Local packet-mark=no-mark limit-at=600k
queue=default priority=8 max-limit=700k burst-limit=0 burst-threshold=0
burst-time=1s

3 name="Network-Up" parent=Local packet-mark=no-mark limit-at=70k
queue=default priority=8 max-limit=80k burst-limit=0 burst-threshold=0
burst-time=1s

4 name="VoIP-control-Up" parent=Phone packet-mark=voip-control limit-at=1M
queue=default priority=3 max-limit=2M burst-limit=0 burst-threshold=0
burst-time=1s

5 name="VoIP-rtp-Up" parent=Phone packet-mark=voip-rtp limit-at=1M
queue=default priority=1 max-limit=2M burst-limit=0 burst-threshold=0

and /ip firewall mangle> print

0 ;;; voip-con-control
chain=prerouting action=mark-connection
new-connection-mark=voip-con-control passthrough=yes
src-address=192.168.200.1-192.168.200.254 dscp=26

1 ;;; voip-con-rtp
chain=prerouting action=mark-connection new-connection-mark=voip-con-rtp
passthrough=yes src-address=192.168.200.1-192.168.200.254 dscp=46

2 ;;; voip-con-control -> mark packet as voip-control
chain=prerouting action=mark-packet new-packet-mark=voip-control
passthrough=yes connection-mark=voip-con-control

3 ;;; voip-con-rtp -> mark packet as voip-rtp
chain=prerouting action=mark-packet new-packet-mark=voip-rtp
passthrough=yes connection-mark=voip-con-rtp

4 ;;; all traffic on LAM mark as: network-con
chain=prerouting action=mark-connection new-connection-mark=network-con
passthrough=yes in-interface=Phone

So far everything I have tried has failed. Link saturates and they cannot hear me. I do not see packets being matched against the rate-limiting for voip packets either which I am sure has to do with it.

Any help greatly appreciated!!

Fearon

Hi. I think the queue root parents look odd.

afaik you need the queue root parent to either be an egress interface or set to global-in/global-out. You'd then have subqueues under these in both directions for phone traffic and other traffic.

So for shaping on egress with two ethernet interfaces on the router you'd have:

Network-Down :parent eth1
--->Voip-Down :parent Network-Down
--->Other-Down : parent Network-Down

Network-Up :parent eth0
--->Voip-Up :parent Network-Up
--->Other-Up :parent Network-Up

If shaping with global in/out the mangle marks should be more specific. Can use in/out interface or in/out bridge port mangle marks to specifically tag up/down on global. There is a network flow diagram around here somewhere that shows you how the router processes packets. Also check out tik-tube which has some good MUM htb/qos presentations.

One other thing is to inspect the traffic and make sure the dscp value is correctly set (or just check the equipment).

Setting the max rate and limit-at looks proper.