Great tool for testing QoS implementation

After my last post, some readers were asking what tool did I used for testing the QoS and how satisfied I’m with it.

The name of the tool is Packgen. According to its developer developers, “Packgen is a simple network packet generator handling diffserv markers, useful for testing network bandwidth and QoS.” It support features like:
– Network packet flows generation with given bandwidth (packets sent at each time interval depending on the bandwidth to produce and the size of the packets to generate);
– UDP and TCP flows;
– DSCP marking;
– Log generation which gives the possibility to compute statistics on the flows (to come later).

The main difference between IPerf (which is also a great testing tool, especially for bandwidth, jitter, packet loss tests…) and Packgen, is that the last one support diffserv marking natively. Of course you can achieve the same results with IPerf, with and ACL and inbound marking of the packets matched in the access-list, but with Packgen this is straight forward.

I had a little issue when downloading this tool, as the first result in Google search engine directed me to a page with no download link: This contains all the needed information about features, installation and how to use it (actually one of the best README that I ever saw for such tools), but no suggestion where to download the package. I search a little bit and you can download it from this official link.

To install this tool, you need to have Ruby on your system and then just run from inside the unpacked Packgen folder:

ruby ./setup.rb

Now some words about how to use it. From my post about AutoQoS, you can see that you need at least one client and one server. This 2 devices, use different files with Packgen (Don’t worry as the files come in the source package and if not, it’s very easy to create them).
First the server file, called listen.yml (if you create it, you can give whatever name you want) looks like this:



ports: !ruby/range 17000..17002

ports: !ruby/range 5002..5004

As you can see, there are 2 sections defined for UDP and TCP traffic. Then with “!ruby/range” you define a range where the server will listen. However, you can also simply use an Integer port number.

Then on the  client side, there is file called sent.yml:



name: Voice
bandwidth: 700Kb
packet_size: 252B
dscp: ef !ruby/range 0.0..60.0

name: Video
bandwidth: 2.8Mb
packet_size: 750B
dscp: cs4 !ruby/range 10.0..60.0

name: Best Effort
bandwidth: 3.2Mb
packet_size: 1KB !ruby/range 20.0..60.0

name: Background
bandwidth: 3.2Mb
packet_size: 1KB
dscp: cs1 !ruby/range 30.0..60.0

Here it’s a little bit more complex, but still human readable. This file also have 2 sections for UDP and TCP traffic, with the following paramters being defined:

-name: I believe it say everything
-host: ServerIP:port
-bandwidth: bandwidth to simulate
-packet size: packet in size in B, KB
-dscp: value !ruby/range: time intervals

After everything is defined, you just have to run Packgen to test.
On Server side first:

packgen -i listen.yml

Then on Client side:

packgen -i sent.yml

Optional you can add the -l file option, which will log the traffic send:

1258663872.6248 SEND dest=
1258663882.62591 SEND dest=
1258663895.65219 SEND dest=
1258663905.66876 SEND dest=
1258663933.72797 STOP dest=

or received:

1258663869.80496 LISTEN port=16384 proto=udp
1258663869.81079 LISTEN port=16385 proto=udp
1258663869.81441 LISTEN port=5002 proto=tcp
1258663869.81506 LISTEN port=5003 proto=tcp
1258663872.5886 RECV sent_at=1258663872.64102 flow=0 size=252 id=6
1258663872.58893 RECV sent_at=1258663872.64613 flow=0 size=252 id=7
1258663872.58903 RECV sent_at=1258663872.64719 flow=0 size=252 id=8
1258663872.58915 RECV sent_at=1258663872.65052 flow=0 size=252 id=9
1258663872.58924 RECV sent_at=1258663872.6742 flow=0 size=252 id=10

The configuration files and logs excerpt were from my AutoQos test. If you have any issues with using it, please contact me, or just check in details the documentation from developers site.

Cisco AutoQoS VoIP

QoS – one of the most interesting and challenging part of the network engineer’s life. I think I’m not wrong when I’m saying that most engineers that hear the word QoS, react somehow like “ah, that ugly stuff…” I have to be honest and say that QoS is not one of my preferred part, but as a network engineer you have to accomplish all task, not only the one that you like. With QoS you can achieve some great results, and once you have started to work with it, you’ll see that is not such a monster. The idea that QoS is not a friendly topic was develop due to historical ways to configure this service. In fact there were so many ways that without working with this topic everyday, you have no chance to remember how has to be implemented. Even among Cisco devices, there were multiple way to configure QoS from device to device. Nowadays Cisco is putting a lot of effort to standardize the implementation of QoS and in the same time to simplify it.

In this context AutoQoS appeared. First, AutoQoS VoIP was developed as the more QoS configuration is needed for Voice traffic over the network and then a more complex Auto QoS Enterprise was released. Today I plan to speak a little bit about AutoQoS VoIP and in one of the future articles about the Enterprise edition.

AutoQoS VoIP gives you the ability to deploy QoS features for converged IP telephony and data networks in a fast and reliable way, by simplifying the MQC (Modular QoS command-line interface). AutoQoS VoIP generate automatically traffic classes and policy map CLI templates. In more human terms, if AutoQoS VoIP is configured on an interface, the traffic that is passing by is receiving the required QoS treatment automatically. By doing this automatically AutoQoS spares you from having in-depth knowledge about the services policies, link  efficiency mechanism or best practice for Voice QoS. Everything is done by AutoQoS, which automatically discovers applications and provides appropriate QoS treatment, generate policies and provide configuration.

Of course that are some drawbacks like in any other technology, but this can be easy overcome if you have some knowledge about QoS configuration. It theory everything should run smoothly, but in real environment, you run into the situation that QoS should be fine tuned to reach the level of your expectation. This is easy done with AutoQoS, as after it enable and configure the QoS templates, you can easy configure them to fine tune parameters, add or delete policies. Basically you start from nice template, sourced by AutoQoS, and develop your own QoS. As I said before, you have only to do this is something is not working or if you feel the strong attraction to “make it your own way”.

AutoQoS VoIP is supported on the most popular Cisco routers and switches platform with the appropriate IOS or CatOS image. Before you configure AutoQoS VoIP you should know that:

– CEF is required to be configured, as AutoQoS use NBAR (Network-Based Application Recognition) which is based on Cisco Express Forwarding
– no QoS Policies are attached on the interface on which you are configuring AutoQoS
– correct bandwidth has to be configured on the interface; if there is no bandwidth statement, then this defaults to 1544 Mbps, which is not all the time accurate

Now a tricky part in regards to interface bandwidth. If this is 768kbps or below, you have to configure an IP address on the interface due to the fact the AutoQoS is enabling by default Multilink PPP and copy the  configured IP under the multilink interface. If you configure AutoQoS on a remote site, and the 768kbps (or lower) interface is the only connection there, please pay attention as configuring AutoQoS Voip may break your connection due to MLP situation explained above.

Another recommendation from Cisco and me, is that you should use no auto qos voip command when you stop using the interface on which AutoQoS VoIP is configured. In this way you assure that all the QoS configuration will be removed. If you just shutdown the interface or delete the PVC, without the above command, then the AutoQoS VoIP will not be completely removed.

Configuration is very simple:

router# configure terminal
router(config)# interface Serial x/y
router(config-if)# bandwidth 1540
router(config-if)# auto qos voip
router(config-if)# exit

I tried to put together a scenario like in the following topology, to generate some traffic and show you have AutoQoS behave when configuring policies:

Cisco AutoQoS VoIP topology

After applying the AutoQoS config you should see something like this in your running-config:

class-map match-any AutoQoS-VoIP-Remark
match ip dscp ef
match ip dscp cs3
match ip dscp af31
class-map match-any AutoQoS-VoIP-Control-UnTrust
match access-group name AutoQoS-VoIP-Control
class-map match-any AutoQoS-VoIP-RTP-UnTrust
match protocol rtp audio
match access-group name AutoQoS-VoIP-RTCP
policy-map AutoQoS-Policy-UnTrust
class AutoQoS-VoIP-RTP-UnTrust
priority percent 70
set dscp ef
class AutoQoS-VoIP-Control-UnTrust
bandwidth percent 5
set dscp af31
class AutoQoS-VoIP-Remark
set dscp default
class class-default
ip access-list extended AutoQoS-VoIP-Control
permit tcp any any eq 1720
permit tcp any any range 11000 11999
permit udp any any eq 2427
permit tcp any any eq 2428
permit tcp any any range 2000 2002
permit udp any any eq 1719
permit udp any any eq 5060
ip access-list extended AutoQoS-VoIP-RTCP
permit udp any any range 16384 32767

I will try to generate some traffic between this, so called, voice server and client and to let AutoQoS  VoIP do it’s job and configure automatically traffic flow policies.

Below you have a video tutorial about AutoQoS VoIP has to be configured and some monitoring commands that will output the result of traffic flow: