I now see that the latest versions of skype dynamically adjust the quality of the video call according to the available bandwidth.

If like me you annoyingly have customers who despite seeing one another at various times of the day, go on to call one another evenings and other times using skype "with video". Why! has always been a mystery, surely you can remember what the person looked like earlier in the day!

So, I set about a method to tame these folk.

The skype video now adjusts its resolution according to "whats available",, with 300k being the minimum and 1.5Mb being the max HD!!!

In the past I had tried various forms of QOS as well as Layer7, all of which seemed "clunky" and often unreliable, so here is a new take:----


Luckily I am blessed with an abundance of skype phones on my network, each with an individual IP.. Each client who uses a phone is stored in an "address-list" with a tag called "phone"

Thus each "phone" user is then allocated a slot within PCQ and a CIR of around 50k.

I noticed that the phones spend most of the day "chattering" across the web, exchanging instants of "whos there?" , "I am", "wheres the server?" etc etc.

Firstly I played with a few mangle rules with careful inclusions/exclusions to catch the destination Ip's so that I could use the data later on in some queues.

The only problem was that it was collecting too much data and I only had confidence in 60% of it, and I ended up with over a thousand entries in the address list within hours.

So, I used a mangle rule to firstly match only the customers who had IP phones (already in the address-list) as "phone"

Then I used connection-rate with 20kb -40kb option satisfy myself that the new entry into the address-list would be more reliable. The so called "chatter" does not have two way contact at those rates, so the reliabilty of identifying skype servers went up to 100%.

Then you can add to the Address-list the details of the destination-address and mark it up as "skype_server"

I then used the new address-list entry to mark the connections and then to mark the packets and change the TOS to 48.
This is all done in the gateway router.

From that point, with a packet mark of "phone" its possible to do various forms of queues to "limit-at" the connection, keeping the bandwidth control down to 300kb, even puting a time against the queue of say 30 minutes was possible.

But better still is the TOS mark of 48 which now passes back into the network upon each valid skype call.

So even at the CPE it would be possible to use just 1 simple queue to collect the TOS mark and keep the bandwidth under control.

Just a few thoughts that I felt I would share