ethernet-default is the type of queue.

Yes, your 3cx server sets the disco value on the packet, but the Mikrotik does not directly see that. You need the mangle rule to make the packet in a way that the router and understand and then place it in the correct queue.

:local wanInterfaceName "Ether1"
:local uploadBandwidth 5500k

:local queueName ("DSCP_" . $wanInterfaceName)
:local dscpClass [:toarray "Network Control,Internetwork Control,Critical,Flash Override,Flash,Immedate,Priority,Routine"]
:local dscp 64

:for thisDscp from 63 to 00 do={
    /ip firewall mangle add action=mark-packet chain=postrouting comment=("dscp_" . $thisDscp) \
         disabled=no dscp=$thisDscp new-packet-mark=("dscp_" . $thisDscp) passthrough=no
}

/queue tree add max-limit=$uploadBandwidth name=$queueName parent=$wanInterfaceName priority=1

:for thisDscp from=0 to=7 do={
   :local subClass ([:pick $dscpClass $thisDscp] . " (" . $wanInterfaceName . ")")
   /queue tree add \ 
      name=($thisDscp+1 . ". " . $subClass ) parent=$queueName priority=($thisDscp+1) queue=ethernet-default
   :for thisQueue from=0 to=7 do={
      :set dscp ($dscp-1)
      /queue tree add name=($subClass . " (Pri: " . $thisQueue+1 .")") \
      parent=($thisDscp+1 . ". " . $subClass) priority=($thisQueue+1) \
      packet-mark=("dscp_" . $dscp) queue=ethernet-default comment=("dscp_" . $dscp)
   }
}

wanInterfaceName needs to match your WAN internet name... normally "ether1", not "Ether1"

I wrote a script to setup DSCP for you.. It's in this thread: http://forum.mikrotik.com/viewtopic.php ... cp#p550055

unable to access the thread/topic, gives me "file not found"?

Can I use this and add the reverse, i.e. use it to prioritize incoming traffic from the Internet

• creation of priority queues for the upload direction
• creation of firewall rules which translate DSCP values to routing marks used to place the packets to appropriate queues

... The reason I ask this question is because I've read that queues only start working when you reach the max limit on the affected traffic. If my internet connection is never near its maximum capacity are queues even necessary?

I know the setup on my VoIP providers side is also a factor. Most likely they do not even have more than 1Gbps breakout but let's assume that they are willing to work with me to set up QoS, what would they need to do?

Second question, I have been considering moving my VoIP server to a hosted environment. I'm planning on connecting my office router to the VoIP server using OpenVPN so that my SIP devices will connect directly to the VoIP server and eliminate the complications that come with NAT'ing VoIP traffic. Does the same apply to VPN connections as per my first question? If not, is their a simple way to protect or prioritise the VPN interface?

I ended up trying out a combination of the script for DSCP above and tweaking it a bit for download as well. I noticed that I can't set my Max Limit to 10G. It seems 1000M is the highest that can be set. That was the reason for me looking up this thread and got me to thinking about the importance of queues.

In terms of our environment, we have about 30 sip devices in a call center and about 10 - 15 active calls at any given time (Tiny, I know XD). Our internet usage hovers around 50 to 100 Mbps during the day with occasional spikes to 200 or 300 Mbps. We haven't had any major complaints about voice quality but I've noticed some audio cut out on occasion. I know if I take this up with my VoIP provider the first thing they will ask is if I have proper QoS in place. I'm not at the stage where I want to investigate call quality yet so I'm just learning a bit more about the proper way to make sure my setup is correct.

In practical terms, jitter values below 100 ms rarely cause voice drop-outs as most VoIP equipment has de-jittering buffers which handle that, but such high values already cause some discomfort as 200 ms round-trip delay is at the edge of acceptability.

In practical terms, jitter values below 100 ms rarely cause voice drop-outs as most VoIP equipment has de-jittering buffers which handle that, but such high values already cause some discomfort as 200 ms round-trip delay is at the edge of acceptability.

How to measure jitter ?

How to measure jitter ?

From my understanding with today’s internet DSCP markings should be preserved through the internet so they can be trusted and/or actioned on the edge devices where needed.