gomez’s blog

There are differences between Call Admission Control mechanisms and Quality of Service. In this blog post, I intend to write about one measurement-based Call Admission Control mechanism and that is IP SLA – which you can use to test end to end Quality of Service within your network. Also, it can be used for measurements which you can use for Advanced Busy Out (AVBO) but in this blog post I will just show how it can be implemented to provide end to end statistics which you can use for detailed analysis for your QoS setup or to test SLA that your service provider needs to meet.

First thing that you need to configure is Responder. That is the router that you will send probes to. If that is your provider and you want to test your link, then you will have to communicate with them to enable this for you. So, from global configuration mode you just need to type:

Router(config)#ip sla monitor responder

and you are done with setting up responder.

Now on your side, you need to configure IP SLA. First step is to define tag for IP SLA (in our case that is number 1) and then we need to define what do we really want to test. In this case it is UDP jitter on voice packages, but you can configure multiple SLAs for different types of traffic depending on your requirements.

VoiceGateway(config)#ip sla 1

Please note that we need to define IP address of the Responder (77.239.X.X) followed by the port (6500) and codec (g711ulaw). Also, codec and port values are adjustable as per your needs.

VoiceGateway(config-ip-sla)#udp-jitter 77.239.X.X 65000 codec g711ulaw

Once done with that, you are able to define Quality of Service value for probe that we defined above, and in our case it is TOS value of 5 which is in decimal 160 (10100000).

VoiceGateway(config-ip-sla-jitter)#tos 160
VoiceGateway(config-ip-sla-jitter)#exit

Once done with that, we need to configure when this probe will be sent and for how long it is going to work, as per following example:

VoiceGatewayconfig)#ip sla schedule 1 start-time now life 180
VoiceGateway(config)#exit

Last thing that we need to do is to verify values:

VoiceGateway#show ip sla statistics

Round Trip Time (RTT) for Index 1
Latest RTT: 1 milliseconds
Latest operation start time: 13:40:03.156 zenica Fri Aug 20 2010
Latest operation return code: OK
RTT Values:
Number Of RTT: 1000 RTT Min/Avg/Max: 1/1/5 milliseconds
Latency one-way time:
Number of Latency one-way Samples: 0
Source to Destination Latency one way Min/Avg/Max: 0/0/0 milliseconds
Destination to Source Latency one way Min/Avg/Max: 0/0/0 milliseconds
Jitter Time:
Number of SD Jitter Samples: 999
Number of DS Jitter Samples: 999
Source to Destination Jitter Min/Avg/Max: 0/1/3 milliseconds
Destination to Source Jitter Min/Avg/Max: 0/1/2 milliseconds
Packet Loss Values:
Loss Source to Destination: 0 Loss Destination to Source: 0
Out Of Sequence: 0 Tail Drop: 0
Packet Late Arrival: 0 Packet Skipped: 0
Voice Score Values:
Calculated Planning Impairment Factor (ICPIF): 1
MOS score: 4.34
Number of successes: 2
Number of failures: 0
Operation time to live: 55 sec

Based on the above output, you are able to see how does your service provider meets their SLA. Also, you are able to see based on the above numbers does your Quality of Service setup really works.

Few days ago I needed to check Daylight Saving Time changes that are going to happen, and I was surprised that there are occasions when you need to update DST settings manually. This example will show how to do that for Egypt. Requirement is to change time on Aug 11, at 5.00AM EEST (Eastern European Summer Time) to 4.00AM EET (Eastern European Time). Here is the procedure how to do that depending of your operating system (Solaris or Linux):

1. Login as user root;
2. Download file tzdata<year>l.tar.gz from ftp://elsie.nci.nih.gov/pub/ or some other tzdata provider:

#wget ftp://elsie.nci.nih.gov/pub/tzdata2010k.tar.gz

3. Extract timezone info file “africa” from the tarball using following command:

#tar -zxf tzdata2010k.tar.gz

#cd tzdata2010k

4. Modify file “africa” or some other zone by using your favorite text editor (vi, nano or something else):

#vi africa

(I have changed it to reflect my needs / EEST will change to EET on Aug 11, 5.00 AM)

5. Depending on your operating system, place file in one of the following directories:

/usr/share/lib/zoneinfo/src – Solaris
/usr/share/zoneinfo/src     – Linux

#mv africa /usr/share/lib/zoneinfo/src

#mv africa /usr/share/zoneinfo/src

6. Navigate to “src” directory and compile a new timezone data file using command “zic”

#cd /usr/share/zoneinfo/src

#zic africa

7. Copy newly generated file

 #cd .. && cp Africa/Cairo Egypt

8. Last step would be to test DST settings with following command:

#zdump -v Egypt | grep 2010

You should be able output similar to this, depending on what you have configured:

#zdump -v Egypt | grep 2010
Egypt Wed Aug 11 09:34:23 2010 UTC = Wed Aug 11 11:34:23 2010 EET isdst=0
Egypt Thu Apr 29 21:59:59 2010 UTC = Thu Apr 29 23:59:59 2010 EET isdst=0
Egypt Thu Apr 29 22:00:00 2010 UTC = Fri Apr 30 01:00:00 2010 EEST isdst=1
Egypt Wed Aug 11 01:59:59 2010 UTC = Wed Aug 11 04:59:59 2010 EEST isdst=1
Egypt Wed Aug 11 02:00:00 2010 UTC = Wed Aug 11 04:00:00 2010 EET isdst=0
Egypt Wed Sep 8 21:59:59 2010 UTC = Wed Sep 8 23:59:59 2010 EET isdst=0
Egypt Wed Sep 8 22:00:00 2010 UTC = Thu Sep 9 01:00:00 2010 EEST isdst=1
Egypt Thu Sep 30 20:59:59 2010 UTC = Thu Sep 30 23:59:59 2010 EEST isdst=1
Egypt Thu Sep 30 21:00:00 2010 UTC = Thu Sep 30 23:00:00 2010 EET isdst=0

That it is it! As you can se from above output time will automatically change on 5.00AM EEST and it will go one hour back to 4.00AM EET. I hope that this blog post clarify this procedure.

1. zic
2. zdump

My previous blog post covers some MGCP foundations and with this blog post I intend to show some other features like MGCP Fallback, PRI Backhaul configuration and DTMF Relay. Occasionally I will post some signaling protocols configurations that I find interesting. Please notice that I will not always post complete solution configuration, it would take additional time and resources but I will point to missing parts. So, lets start with “MGCP Part 2″.

MGCP Fallback

Let’s assume that you have configured your Cisco Unified Communications Manager to control your Cisco Voice gateway as described in my previous blog post. In that case, you will have your gateway fully controlled by MGCP and it is going to work… until there is a WAN failure or some other communication issue as shown on following picture.

As shown on above picture, in case of WAN failure we need some kind of fall-back to assure that we have our business ongoing. Telephony is definitely one of the mission critical services which needs different solutions in order to achieve normal business continuity and one of those solutions is MGCP Fallback. Basically, as shown on above picture we need to make gateway to fallback to some other signaling protocol than MGCP in case of WAN failure or some other failure that could cause problems with connection between gateway and Unified Communications Manager. By default, fallback protocol of choice in this case is H.323. So basically, to achieve that, we need to preform following configuration:

Router(config)#application
Router(config-app)#global
Router(config-app-global)#service alternate Default

Once we are done with configuring MGCP Fallback, we would need to configure complete dial plan for H.323 by creating dial peers for each destination/type of call (for example: national or international) in order to be able to establish a call. It is not my intention to cover that part, since I assume that you already configured it.

To verify, we would need to execute following commands:

Router#show ccm-manager
Router#show ccm-manager fallback-mgcp

PRI/BRI Backhaul Configuration

MGCP PRI/BRI Backhaul is mostly used when we are bridging Unified Communications Manager with some third party PBX using QSIG. While talking with networking (not telco) guys I would describe it this way: L2 signaling is maintained by voice gateway and L3 signaling is passed to Cisco Unified Communications Manager by voice gateway.

MGCP PRI backhaul is a method for transporting complete IP telephony signaling information from an ISDN PRI interface in an MGCP gateway to Cisco Unified Communications Manager using a highly reliable TCP connection. The gateway uses a single TCP connection to backhaul all ISDN D channels to Cisco Unified Communications Manager. MGCP PRI backhaul terminates all ISDN PRI Layer 2 (Q.921) signaling functions on the MGCP gateway while, at the same time, packaging all the ISDN PRI Layer 3 (Q.931) signaling information into packets for transmission to Cisco Unified Communications Manager through an IP tunnel over a TCP connection.

So, to configure PRI/BRI Backhaul I will assume that your E1/T1 controller is already configured with base configuration (line code, clocking, framing).

Router(config)#isdn switch-type primary-net5
Router(config)#controller E1 1/0
Router(config-controller)#pri-group timeslots 1-32 service MGCP
Router(config-controller)#exit
Router(config)#interface serial 1/0:16
Router(config-if)#isdn bind-l3 ccm-manager

So basically, we defined our isdn-switch type globally (we could do that on controller level as well) and after that we started configuring our controller. We created pri-group which is basically controlled by MGCP. Once we applied that, we need to switch to serial interface created by execution of previous command and we need to instruct gateway to transfer all Q.931 signaling to Unified Communications Manager.

Last thing that we need to do is to verify above configuration and we can do that by executing following commands:

Router#show isdn status
Router#show ccm-manager backhaul

DTMF Relay Configuration

If you have some services within your organization that require DTMF functions (good example would be some IVR implementation for support where customer is typing incident number in order to be connected to engineer that is working on incident) and you are facing some difficulties with that (not all digits are received and so on) solution would be to extract that signaling from codec bandwidth and process it out-of-band.

To achieve that, we would need following configuration:

Router#conf t
Router(config)#mgcp dtmf-relay voip codec all mode out-of-band

Once done with that, don’t forget to save your configuration.

For more information, refer to following links:

1. MGCP Fallback
2. DTMF Relay
3. PRI/BRI Backhaul

In following blog posts I will write about different Cisco based VoIP solutions and I will try to cover some foundation topics like voice signaling protocols and so on. My aim is to present different solutions and possibilities and I hope that you will find it interesting and helpful.

About MGCP and reasons to use it

Media Gateway Control Protocol is client server voice signaling protocol. Call control is handled by call agent (Cisco Unified Communications Manager) and media translation is performed by voice gateway. It is important to mention that it is only client – server model based voice signaling protocol on the market. Also, it is well know for its simple configuration (it is just matter of configuring call agent and remembering several gateway commands), centralized management and one of the main benefits is it’s use for MGCP back-haul QSIG configurations (in short: Layer 2 signalization is handled by voice gateway which is under control of call agent, and layer 3 signalization is forwarded to call agent – Cisco Unified Communications Manager – CallManager). Please note that MGCP is often called Megaco by telco people.

MGCP Messages

1. Create connection – CRCX
2. Notification request – RQNT
3. Notify – NTFY
4. Modify connection – MDCX
5. Delete connection – DLCX
6. Audit endpoint – AUEP
7. Endpoint configuration – EPCF
8. Restart in progress – RSIP

MGCP Call flow

MGCP configuration basics

This part of text will cover MGCP configuration – just foundations. Please note that it won’t include all specific configurations, it will just cover base configuration, but in several new posts I will try to cover different configuration scenarios and solutions.

So, let assume that MGCP call agent, in our case Cisco Unified Communications Manager – CallManager is working on IP address 192.168.1.95. Also, assume that redundant CallManager is working on IP address 192.168.1.100.  This text does not show CallManager side of configuration of a gateway and I assume that that part is already successfully configured (I will eventually edit this blog post to include that configuration, afterwards)

Configurations is as follows:

1. From global configuration mode start MGCP proces:

Router#conf t
Enter configuration commands, one per line.  End with CNTL/Z.
Router(config)#mgcp
Router(config)#

2. Next thing we need to do is to configure call agent, redundant call agent and we need to inform voice gateway that call agent is under control of MGCP. We can do that as follows:

Router(config)#mgcp call-agent 192.168.1.95
Router(config)#ccm-manager redundant-host 192.168.1.100
Router(config)#ccm-manager mgcp

3. Once done with that, we can configure out Cisco Unified Communications Manager gateway options. So to do that, navigate to your CallManager and in Cisco Unified CM Administration configuration menu select Device > Gateway > Add new. Select your voice gateway model from the drop down list and click Next. For Protocol select MGCP and click Next. Once there, configure Domain name (please note that this field needs to be in following format: gatewayhostname.domainname, for example CCIEVoiceLab.localnetwork.lab) and other required options (add cards and preform complete gateway configuration). Once you are done with voice gateway configuration you are required to configure dial plan which will include this gateway into configuration. That goes beyond the scope of this blog post and I will write about it in some of following blog posts.

4. Once done with CallManager configuration we need to instruct voice gateway about the TFTP address where CallManager stores configuration, and that is typically IP address of the CallManager it self:

Router(config)#ccm-manager config server 192.168.1.95
Router(config)#

5. Once we are done with that we can instruct our gateway to download and apply configuration by issuing following command:

Router(config)#ccm-manager config
Router(config)#

Wait several seconds and you can verify your downloaded and applied configuration with following command:

Router#show running-config
Router(config)#

At that point your voice gateway will be controlled by call agent, in this case CallManager. To verify your configuration please give a check to Troubleshooting commands.

MGCP Troubleshooting

Once you are done with MGCP gateway configuration, you can verify your configuration by issuing following commands:

Router#show mgcp
MGCP Admin State ACTIVE, Oper State ACTIVE – Cause Code NONE
MGCP call-agent: 192.168.1.95 Initial protocol service is MGCP 0.1
MGCP validate call-agent source-ipaddr DISABLED
MGCP validate domain name DISABLED
MGCP block-newcalls DISABLED
MGCP send SGCP RSIP: forced/restart/graceful/disconnected DISABLED
…

This command will show all MGCP settings, timers and so on.

Router#show ccm-manager
MGCP Domain Name: Router
Priority        Status                   Host
======================================
Primary         Registering with CM      192.168.1.95
First Backup    Down                     192.168.1.100
Second Backup   None

Current active Call Manager:    None
Backhaul/Redundant link port:   2428
…

This command will show registration status and other statistics.

Beside above commands, we can use:

Router#show mgcp endpoint
Router#show voice port summary

That would be all for this post. I hope that this blog post is helpful and if you do have some questions in regards to MGCP please let me know and we can try to find solution. In next several posts I will write about MGCP fallback, backhaul and DTMF configurations.

1. Media Gateway Control Protocol
2. MGCP messages

Two years ago I have decided to start working on Cisco Voice Professional certification track. It was logical step to make for me since I was working for several years in voice industry with different solutions, so when I started with Cisco solutions there was no doubt what to choose and why. In prior to further discussion about certification I would like to point that I am full time employed with lot of different responsibilities which can and does impact preparations for certification.  So, for some people that might take this certification track this might be resolved faster, but all depends on attitude, basic knowledge and many other factors. I took a newer track based and created around Cisco Unified Communications Manager 6.0. When I started preparing myself for CCVP, there was active certification based on Cisco Unified Communications Manager 4.1, and main difference beside versions of Cisco Unified Communications Manager  is that old certification path included Gateway and Gatekeeper exam. I took all of the exams as Cisco is recommending on their web site for CCVP certification. So, that is as follows:

Cisco Certified Network Associate Voice – CCNA V

For all English language readers of this blog, this is blog post in Bosnian language about opening of a new Pearson VUE testing centar in Zenica, Bosnia and Herzegovina at REZ agency.

Cisco certifikati su jedni od najcjenjenijih referensi koji su dostupni stručnjacima u oblasti informacione tehnologije. Stoga je, kao logičan završetak obuke za polaznike Cisco Akademije, REZ Agencija u svom prostoru uspostavila testni centar najpoznatijeg testing provajdera Pearson Virtual Universities Enterprises (Pearson VUE), prema veoma strogim tehničkim sigurnosnim kriterijima.

Usluge Pearson VUE testnog centra koriste mnoge poznate kompanije i organzacije kao što su Cisco, Linux Professional Insitute, CompTIA i drugi. Za ispite u Pearson VUE testnom centru u REZ Agenciji se možete prijaviti direktno preko weba http://www.vue.com/, putem telefona 032 441 231 ili dolaskom u našu Agenciju kod PVTC Administratora Testnog centra.

Plaćanje za Vaše ispite možete vršiti direktno putem stranice http://www.vue.com ili se obratiti administratoru u našem testnom centru (gđa Mediha Zukić).

1. REZ Agencija
2. Pearson VUE

Few days ago I moved my Ubuntu on laptop to version 10.04 and so far I am very satisfied with how does it work. That was a clear sign to refresh some software installs that I have and that fact included VMware server. Basically, I found VMware to suit my needs when it comes to personal virtualization. I am using virtualized CallManager (it is only supported with VMware) and also my appliance Linux environment is virtualized as well. There are other virtualisation solutions that I like but I will leave that for some other blog post.

So, I installed VMware Server 2.0.2 and it is working really nice. There are plenty of instruction manuals on VMware installation on Linux and Ubuntu, but I found this one to work like a charm. Under Ubuntu, I am using current version of Firefox, which is 3.6.3 and it seems to have a problem with VMware Remote Console Plug-in. It is not working at all. There is lot of discussion about this but I didn’t find that much solutions. Some are saying that some Firefox add-on for flash blocking is causing this to happen, but using Firefox without .mozilla didn’t help.

So, here is solution that actually works: Open Terminal and navigate to directory:

root@gomez-laptop:/# cd /usr/lib/vmware/webAccess/tomcat/apache-tomcat-6.0.16/webapps/ui/plugin/

once there, execute list command and you should be able to see vmware-vmrc-linux-x86.xpi file which is required:

root@gomez-laptop:/usr/lib/vmware/webAccess/tomcat/apache-tomcat-6.0.16/webapps/ui/plugin# ls
build_doNotErase.txt  vmware-vmrc-linux-x64.xpi  vmware-vmrc-linux-x86.xpi vmware-vmrc-win32-x86.exe  vmware-vmrc-win32-x86.xpi

Once, there, we need to unzip that file by issuing following command:

root@gomez-laptop:# unzip vmware-vmrc-linux-x86.xpi

and then, navigate to plugins directory and list the content:

root@gomez-laptop:/usr/lib/vmware/webAccess/tomcat/apache-tomcat-6.0.16/webapps/ui/plugin# cd plugins/
root@gomez-laptop:/usr/lib/vmware/webAccess/tomcat/apache-tomcat-6.0.16/webapps/ui/plugin/plugins# ls
bin  libconf                         open_source_licenses.txt  vmware-desktop-entry-creator  vmware-vmrc-daemon  xkeymap
lib  np-vmware-vmrc-2.5.0-122581.so  share                     vmware-vmrc vmware-vmrc-legacy
root@gomez-laptop:/usr/lib/vmware/webAccess/tomcat/apache-tomcat-6.0.16/webapps/ui/plugin/plugins#

And one last step is to start vmware-vmrc with following command:

root@gomez-laptop:/usr/lib/vmware/webAccess/tomcat/apache-tomcat-6.0.16/webapps/ui/plugin/plugins# ./vmware-vmrc -h localhost:8333

Please note that you can replace localhost with IP address of your working VMware server. Once above command is executed you need to authenticate and you should be able to select virtualized hosts. Please give a check to following screenshots:

As shown on above screenshot, we need to authenticate and one last step is to select one of the virtualized machines that we want to work with:

If there are some questions, please let me know. I hope that this workaround will save some time for you and that you will have working solution.

1. VMware server
2. VMware server installation on Ubuntu

Richard M. Stallman is an American software freedom activist and computer programmer. In September 1983, he launched the GNU Project to create a free Unix-like operating system, and has been the project’s lead architect and organizer. With the launch of the GNU Project, he initiated the free software movement and, in October 1985, set up the Free Software Foundation.

Stallman pioneered the concept of copyleft and he is the main author of several copyleft licenses including the GNU General Public License, the most widely used free software license. Since the mid-1990s, Stallman has spent most of his time advocating for free software, as well as campaigning against both software patents and what he sees as excessive extension of copyright laws. Stallman has also developed a number of pieces of widely-used software, including the original Emacs, the GNU Compiler Collection, and the GNU Debugger. He co-founded the League for Programming Freedom in 1989.

Richard M. Stallman will have following speeches in Bosnia and Hercegovina in May, 2010:

What: Copyright vs. Community
When: May 19, 2010 from 11:00 AM to 01:30 PM
Where: IBU conference hall, main building, third floor, Francuske revolucije bb; Sarajevo, Bosnia and Herzegovina

What: Free Software Movement and the GNU/Linux Operating System
When: May 20, 2010 from 02:00 PM to 04:30 PM
Where: IBU conference hall, main building, third floor, Francuske revolucije bb; Sarajevo, Bosnia and Herzegovina

What: Free Software Movement and the GNU/Linux Operating System
When: May 22, 2010 from 14:00 PM to 16:00 PM
Where: Fakultet Informacijskih Tehnologija, Univerzitet Dzemal Bijedic; Mostar, Bosnia and Herzegovina

We hope that you will come to listen speeches from Stallman and that you will enjoy those events. It is good opportunity to visit those events since we do not know when we will have Stallman in Bosnia and Hercegovina again. If there is additional information needed, please let me know.

1. Richard M. Stallman
2. Free Software Foundation

Few months ago I passed Implementing Cisco Quality of Service (QoS). It was one of those “real life” exams with lot of simulations and drag and drop questions. Passing score was set to around ~80% and there was around 50 questions or so. Since Quality of Service is one of the huge areas there is lot of viable documentation. I would recommend Cisco QOS Exam Certification Guide (IP Telephony Self-Study), 2nd Edition from Cisco Press, but there are plenty of other books focused on this area as well. One of the best organized online learn sessions that I have found to be focused on this area was written by Paul Stryer from Global Knowledge. Please find links below text pointing to that online learn session. I need to mention that it is best practice to give a try to all of the solutions that you can find in this texts on real equipment, deploy Quality of Service in your organization and you will gain required understanding and experience to pass this exam. Beside that, you will have your network working much better than it was in prior to implementing QoS within your organization. Since I am following Cisco CCVP track this was only exam that is not directly related to voice (I mean it is not organized around voice strictly) and it was very nice experience. Personally I really enjoyed learning and preparing this stuff, and most of the solutions that I have learned are something that needs and can be implemented in networks that I was working with.

Quality of Service, Part 1 – Introduction
Quality of Service, Part 2 – Introduction
Quality of Service, Part 3 – Introduction
Quality of Service, Part 4 – QoS Mechanisms
Quality of Service, Part 5 – Classification
Quality of Service, Part 6 – Marking
Quality of Service, Part 7 – Service Policy
Quality of Service, Part 8 – Congestion Management
Quality of Service, Part 9 – FIFO Queuing
Quality of Service, Part 10 – Weighted Fair Queuing
Quality of Service, Part 11 – CBWFQ
Quality of Service, Part 12 – Low Latency Queuing
Quality of Service, Part 13 – MQC Pop Quiz
Quality of Service, Part 14 – MQC Pop Quiz Answer

I hope that you will enjoy reading above texts and that you will enjoy implementing Cisco Quality of Service. If you would need some additional information please leave a comment.

Few days ago I had interesting request to tweak Linux on Samsung LCD TV. At first, I was kind of surprised that Linux is supported on TV, but it is normal. Linux is very strong competitor on the appliance based market and it was just about time to face those kind of installations. The main goal was to enable telnet on firmware in order to be able to connect to TV, and after that to load Common Internet File System to be able to use network mounts. Since I didn’t know anything about Samsung TV series and belonging firmware versions I needed to read something about that. Basically, there are differences in versions based on producing years and processors that are supported. This text is about to cover Samsung LCD TV LE40A756 and A856 with firmware T-RBYDEU. I need to mention that it is kind of easier to work with other models of Samsung LCD TVs since they seam to open more possibilities (this is just my opinion) for firmware hacking but this version was very interesting to play with as well. Since this firmware (as much as all other appliance based firmware versions) is using squash file system, to be able to add something on filesystem we would need to unsquash file system (unsquash is viable thru squashfs-tools) and then perform necessary changes. Since I am using Ubuntu which is providing newer version of squashfs-tools package than it is required (Ubuntu is providing squashfs-tools version 3.1 and it is required to have version 2.0) I didn’t follow that procedure. However, for those that would like to give it a try you can follow this procedure. We will follow other procedure.

Downloads

Navigate to Samsung web site > Support > Downloads > TV Audio Video > Choose a product type>: Television > Choose a product subtype: LCD TV > Select your TV model and then click on Select. You will be provided with the link to download your firmware version. In our case firmware is called T-RBYDEU.exe.

Firmware work

Using some of the decompression tools like 7zip, WinRAR or some other program that you use on the daily basis, we need to extract .exe file that we have downloaded. I will describe procedure under Linux operating system using 7zip:

7z x T-RBYDEU.exe

Once done, you will be able to see extracted files in directory T-RBYDEU. Please note that you will be able to see those files:

gomez@gomez-laptop:/media/Data_/T-RBYDEU$ ls -la
total 40
drwx—— 1 gomez gomez     0 2009-05-11 23:38 .
drwx—— 1 gomez gomez  4096 2010-04-01 16:22 ..
-rwxrwxrwx 1 gomez gomez 18929 2009-04-17 16:16 crc
drwx—— 1 gomez gomez  4096 2009-05-11 23:38 image
-rwxrwxrwx 2 gomez gomez  8676 2009-04-17 16:16 MicomCtrl
gomez@gomez-laptop:/media/Data_/T-RBYDEU$ cd image/
gomez@gomez-laptop:/media/Data_/T-RBYDEU/image$ ls -la
total 58166
drwx—— 1 gomez gomez     4096 2009-05-11 23:38 .
drwx—— 1 gomez gomez        0 2009-05-11 23:38 ..
-rwxrwxrwx 1 gomez gomez  8155136 2009-04-17 16:16 appdata.img
-rwxrwxrwx 1 gomez gomez   602112 2009-04-17 16:16 boot.img
-rwxrwxrwx 1 gomez gomez 37646336 2009-04-17 16:16 exe.img
-rwxrwxrwx 1 gomez gomez   177920 2009-04-17 16:16 fnw.bin
-rwxrwxrwx 1 gomez gomez       17 2009-04-17 16:16 info.txt
-rwxrwxrwx 1 gomez gomez    99680 2009-04-17 16:16 oneboot.bin
-rwxrwxrwx 1 gomez gomez  3424256 2009-04-17 16:16 rootfs.img
-rwxrwxrwx 2 gomez gomez  3172612 2009-04-17 16:16 T-RBYDAUM
-rwxrwxrwx 2 gomez gomez  3172612 2009-04-17 16:16 T-RBYDEUM
-rwxrwxrwx 1 gomez gomez   228296 2009-04-17 16:16 u-boot.bin
-rwxrwxrwx 2 gomez gomez       47 2009-04-17 16:16 validinfo.txt
-rwxrwxrwx 2 gomez gomez       43 2009-04-17 16:16 version_info.txt
-rwxrwxrwx 1 gomez gomez  2864533 2009-04-17 16:16 vmlinux.ub
gomez@gomez-laptop:/media/Data_/T-RBYDEU/image$

As per above output, please note that we have directory called image, and from the interesting files we have exe.img and validinfo.txt which will be used in following text.

Firmware tweaks

Next thing that we need to do is to prepare exe.img for additional changes in order to be able to telnet to TV and use some networking services like CIFS. Now, navigate to image directory and create directory temp:

gomez@gomez-laptop:/media/Data_/T-RBYDEU$ cd image/
gomez@gomez-laptop:/media/Data_/T-RBYDEU/image$ mkdir temp
gomez@gomez-laptop:/media/Data_/T-RBYDEU/image$

Once done, we will mount exe.img with the type of vfat to newly created temp directory:

gomez@gomez-laptop:/media/Data_/T-RBYDEU/image$ sudo mount -t vfat -o loop exe.img temp/

With above command, we have mounted exe.img to temp directory. Please note that we have following files inside temp directory:

gomez@gomez-laptop:/media/Data_/T-RBYDEU/image$ cd temp/
gomez@gomez-laptop:/media/Data_/T-RBYDEU/image/temp$ ls -la
total 28980
drwxr-xr-x 3 root root 16384 1970-01-01 01:00 .
drwx—— 1 gomez gomez 4096 2010-04-04 19:06 ..
-rwxr-xr-x 1 root root 29487160 2009-04-17 16:16 exeDSP
drwxr-xr-x 3 root root 8192 2009-04-17 16:16 modules
-r-xr-xr-x 1 root root 131072 2009-04-17 16:16 $rfs_log.lo$
-r-xr-xr-x 1 root root 8192 2009-04-17 16:16 rfs_pool.sy$
-rwxr-xr-x 1 root root 4374 2009-04-17 16:16 run
-rwxr-xr-x 1 root root 4525 2009-04-17 16:16 start.sh
gomez@gomez-laptop:/media/Data_/T-RBYDEU/image/temp$

Please note that we have start.sh file which we will use in following text.

Scripts

Using your favourite text editor, replace current start.sh file with following:

#!/bin/sh

if [ -e /mtd_chmap/start.sh ]
then
echo “user start.sh found!”
/mtd_chmap/start.sh
else

export PATH=/sbin:/usr/sbin:/usr/bin:/bin:/usr/local/bin:/usr/local/sbin
#export LD_LIBRARY_PATH=/usr/local/lib:/usr/lib:/lib
export LD_LIBRARY_PATH=$LD_LIBRARY_PATH:/mtd_cmmlib:/usr/local/lib:/usr/lib:/lib
export HOME=/tmp/root
export TERM=linux
export LOGNAME=”root”
cd /dtv/usb
PU=`cat log | grep Mount`
PU=`echo ${PU##MountDir : }`
echo $PU
$PU/usb.sh &

mount

## mount -n -t usbfs none /proc/bus/usb

ulimit -s 2048

export PS1=’\h:\w\$ ‘

## STAPI Root Device Name?## ———————-
ST_DEV_ROOT_NAME=stapi
export ST_DEV_ROOT_NAME

## STAVMEM Device Name
## ——————-
STAVMEM_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stavmem_ioctl
export STAVMEM_IOCTL_DEV_PATH

## STAUDLX Device Name
## ——————-
STAUDLX_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/staudlx_ioctl
export STAUDLX_IOCTL_DEV_PATH

## STBLAST Device Name
## ——————-
STBLAST_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stblast_ioctl
export STBLAST_IOCTL_DEV_PATH

## STBLIT Device Name
## ——————
STBLIT_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stblit_ioctl
export STBLIT_IOCTL_DEV_PATH

## STCC Device Name
## —————-
STCC_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stcc_ioctl
export STCC_IOCTL_DEV_PATH

## STCLKRV Device Name
## ——————-
STCLKRV_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stclkrv_ioctl
export STCLKRV_IOCTL_DEV_PATH

## STDENC Device Name
## ——————
STDENC_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stdenc_ioctl
export STDENC_IOCTL_DEV_PATH

## STEVT Device Name
## —————–
STEVT_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stevt_ioctl
export STEVT_IOCTL_DEV_PATH

## STFASTFILTER Device Name
## ————————
STFASTFILTER_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stfastfilter_ioctl
export STFASTFILTER_IOCTL_DEV_PATH

## STFDMA Device Name
## ——————
STFDMA_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stfdma_ioctl
export STFDMA_IOCTL_DEV_PATH

## STGFB Device Name
## —————–
STGFB_CORE_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stgfb_core
export STGFB_CORE_DEV_PATH
STGFB_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stgfb_ioctl
export STGFB_IOCTL_DEV_PATH

## STHDMI Device Name
## ——————
STHDMI_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/sthdmi_ioctl
export STHDMI_IOCTL_DEV_PATH

## STI2C Device Name
## —————–
STI2C_IOCTL_DEV_PATH=”/dev/${ST_DEV_ROOT_NAME}/sti2c_ioctl”
export STI2C_IOCTL_DEV_PATH

## STLAYER Device Name
## ——————-
STLAYER_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stlayer_ioctl
export STLAYER_IOCTL_DEV_PATH

## STMERGE Device Name
## ——————-
STMERGE_IOCTL_DEV_PATH=”/dev/${ST_DEV_ROOT_NAME}/stmerge_ioctl”
export STMERGE_IOCTL_DEV_PATH

## STPIO Device Name
## —————–
STPIO_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stpio_ioctl
export STPIO_IOCTL_DEV_PATH

## STPTI4 Device Name
## ——————
STPTI4_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stpti4_ioctl
export STPTI4_IOCTL_DEV_PATH

## STSMART Device Name
## ——————-
STSMART_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stsmart_ioctl
export STSMART_IOCTL_DEV_PATH

## STSYS Device Name
## —————–
STSYS_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stsys_ioctl
export STSYS_IOCTL_DEV_PATH

## STTTX Device Name
## —————–
STTTX_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stttx_ioctl
export STTTX_IOCTL_DEV_PATH

## STTUNER Device Name
## ——————-
STTUNER_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/sttuner_ioctl
export STTUNER_DEV_PATH

## STVBI Device Name
## —————–
STVBI_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stvbi_ioctl
export STVBI_IOCTL_DEV_PATH

## STVID Device Name
## —————–
STVID_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stvid_ioctl
export STVID_IOCTL_DEV_PATH

## STVIN Device Name
## —————–
STVIN_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stvin_ioctl
export STVIN_IOCTL_DEV_PATH

## STVMIX Device Name
## ——————
STVMIX_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stvmix_ioctl
export STVMIX_IOCTL_DEV_PATH

## STVOUT Device Name
## ——————
STVOUT_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stvout_ioctl
export STVOUT_IOCTL_DEV_PATH

## STVTG Device Name
## —————–
STVTG_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/stvtg_ioctl
export STVTG_IOCTL_DEV_PATH

STTKDMA_IOCTL_DEV_PATH=/dev/${ST_DEV_ROOT_NAME}/sttkdma_ioctl
export STTKDMA_IOCTL_DEV_PATH

RUN=1
/exe/exeDSP

Please note bold lines. Basically, it is the same file except that we are instructing our Linux firmware to load usb.sh script which needs to be located on USB root directory. usb.sh script will instruct Linux firmware to start busybox instance of telnet deamon so that we are able to telnet to our TV, and additionally it will load cifs.ko kernel module which will enable Common Internet File System so that we are able to mount some network devices on our Linux firmware. usb.sh script looks like:

#! /bin/sh

echo “!!!!!!!!!!!!!! USB START !!!!!!!!!!!!!!”

/bin/stty -F /dev/ttyAS0 ospeed 57600 ispeed 57600
#line before enables console for speed 57600. Put ‘debug’ to console and you see menu, put ’12′ and ENTER
echo “!!!!!!!!!!!!!! start console on 57600 !!!!!!!!!!!!!!”

#sleep 25 sec is need for wait ethernet start
sleep 25

cd /dtv/usb
PU=`cat log | grep Mount`
PU=`echo ${PU##MountDir : }`

echo “!!!!!!!!!!!!!! start telnet !!!!!!!!!!!!!!”
$PU/busybox telnetd -l /bin/sh

echo “!!!!!!!!!!!!!! start cifs.ko !!!!!!!!!!!!!!”
cd $PU
insmod cifs.ko

sleep 4

echo “!!!!!!!!!!!!!! clear old contents USB on TV !!!!!!!!!!!!!!”
rm /mtd_contents/V* -R
rm /mtd_contents/database -R

echo “!!!!!!!!!!!!!! start mount !!!!!!!!!!!!!!”
mount -t cifs //192.168.0.1/photo $PU/photo -o user=user_xp -o pass=password_xp
mount -t cifs //192.168.0.1/mp3 $PU/mp3 -o user=user_xp -o pass=password_xp
mount -t cifs //192.168.0.1/video $PU/video -o user=user_xp -o pass=password_xp

mount

echo “!!!!!!!!!!!!!! USB END !!!!!!!!!!!!!!”

Please note bold lines. Basically, we are starting telnet daemon and loading cifs.ko kernel module. After that, we are mounting some Windows XP shares from our personal computer which is working on 192.168.0.1 IP address. Once this file is loaded, and those mount locations are created you will be able to use files within above directories on your TV from the TV menus.

Please note that usb.sh, cifs.ko, busybox files and mp3, video, and photo directories needs to be located on USB root location.

Once done with that, we are ready to preform final changes to our firmware. First thing that we need to do is to umount temp directory:

gomez@gomez-laptop:/media/Data_/T-RBYDEU$ sudo umount /home/gomez/Desktop/T-RBYDEU/temp

Then, we need to delete temp directory:

gomez@gomez-laptop:/media/Data_/T-RBYDEU/image$ sudo rm -rf temp/

Once done with that, we need to change CRC information, because we have changed exe.img file. We can do that as follows:

gomez@gomez-laptop:/media/Data_/T-RBYDEU/image$ crc32 exe.img
cdc8080c

Once we have CRC information, we need to enter new value to validinfo.txt file:

gomez@gomez-laptop:/media/Data_/T-RBYDEU/image$ cat validinfo.txt
*007_exe.img_663430d9*011_appdata.img_a485b538

Please note value 663430d9. We need to change that value with cdc8080c that we have got as output from crc32 command. You can do that by using your favourite text editor. Once done with that, we are done with tweaking our Linux firmware. Your USB pen content should look like this:

gomez@gomez-laptop:~/Desktop/USB$ ls -la
total 3644
drwxr-xr-x 6 gomez gomez    4096 2010-04-05 08:59 .
drwx—— 8 gomez gomez    4096 2010-04-05 09:03 ..
-rwxrwxrwx 1 gomez gomez  992084 2009-11-18 23:13 busybox
-rwxrwxrwx 1 gomez gomez 2707316 2009-11-17 20:08 cifs.ko
drwx—— 2 gomez gomez    4096 2009-12-28 09:12 mp3
drwx—— 2 gomez gomez    4096 2009-12-28 09:12 photo
drwx—— 3 gomez gomez    4096 2010-04-05 08:59 T-RBYDEU
-rwxrwxrwx 1 gomez gomez    1026 2009-12-28 09:14 usb.sh
drwx—— 2 gomez gomez    4096 2009-12-28 09:12 video

If that is the case, you can put this USB to your TV, turn it on and you will be prompted to upgrade firmware with firmware version listed on your USB pen. Once you perform this upgrade, you will be able to telnet to your TV. Check your TV IP address from TV menus, and then using Linux shell, or if you are Windows user using Putty, telnet to your TV and enjoy power of your Linux shell. Also, using TV menus and Common Internet File System – CIFS, you can play some music, video or photos on your TV (as described above).

If you are lazy to give a check to this procedure, you can download fully operational USB image on this link. You are doing that on your own risk and author of this blog post is not responsible for any damage that you might cause by some operation.

Enjoy power of Linux shell on your Samsung TV!

For more information please visit following links:

1. SamyGo wiki pages
2. How to enable Telnet on Samsung TV’s
3. Hacking T-RBYDEUC Firmware – LE40A756 and A856

LUGZDK finally has new logo. Logo is very simple and well designed, as per our wishes. Man that stands behind the project is Alem Sacak, experienced designer from Zenica, Bosnia and Hercegovina. We would like to thank Alem for this really good job and we are looking forward to work with him in the future on similar projects. If you are interested in his work, you can find his graphic design portfolio and contact details on this link.

1. Alem Sacak
2. Linux users group of ZDK

XENOPHONIA, the project of Jazz Fest Sarajevo is planned as an annual musical event whose aim is to contribute to improvement of music production in the region through improving cross-border cooperation, a better flow and quality of information and promotions, as well as the development of production in the artistic and technical sense.

08.04.2010. 20:00

Türkan, Zulfikarpašić & Garcia-Fons: SEVDALINKAS

Turska, BiH, Španija

Bosanski kulturni centar, Branilaca Sarajeva 24

08.04.2010. 22:15

SUPERSILENT
Norveška

stage

09.04.2010. 20:00

Arve Henriksen
Norveška

stage

10.04.2010. 16:00

Talk with Nils Petter Molvær
Norveška

Bosanski kulturni centar, Branilaca Sarajeva 24

10.04.2010. 19:00

BUZZ
BiH, Austrija, Njemačka

stage

10.04.2010. 21:00

Nils Petter Molvær
Norveška

Bosanski kulturni centar, Branilaca Sarajeva 24

11.04.2010. 12:00

Koncert za djecu
Bosna i Hercegovina

Bosanski kulturni centar, Branilaca Sarajeva 24

11.04.2010. 13:00

Radionica za djecu
Bosna i Hercegovina

Bosanski kulturni centar, Branilaca Sarajeva 24

For more information about Xenophonia visit following links:

1. Xenophonia: About
2. Xenophonia: Timeline

I was surprised when I tried to place a VoIP call over UMTS/3G network using my Nokia E72. It wasn’t working. In prior to that, I needed to install Nokia SIP VoIP settings application to get VoIP support on my mobile phone (Nokia E71 comes with this software preinstalled, so if you are Nokia E71 user and possible switcher to Nokia E72 be aware of this). I was really positively surprised when I saw how many different options can you configure using Nokia SIP VoIP settings application (for example, you can change your QoS DSCP value for RTP traffic, RTP port range, codecs and so on… impressive.) However, my VoIP calls placed over 3G were unsuccessful. So, I needed to find solution for it.

As a prerequisite make sure that you are connected to UMTS network by navigating to:

Home > Control Panel > Settings > Phone > Network > Network Mode

While there for “Network Mode” field you will need to set “UMTS” and once you apply that you will be connected to your UMTS/3G network (in upper left corner you will be able to see small 3G connection mark).

Once done with that we need to enable 3G VoIP calls.

Procedure is as follows: Navigate to Home > Control Panel > Net Sett:

Once there, navigate to “Advanced VoIP settings” and then select “VoIP services“:

After that you will need to select your SIP profile (which is in our case called devlogic) and after that select “Profile settings” as per screenshot:

Once there, you will need to scroll down and find option “AWCDMA“. That option is set to “off” by default, and you will need to set it to “on” in order to place VoIP calls over 3G network.
W-CDMA (Wideband Code Division Multiple Access), UMTS-FDD, UTRA-FDD, or IMT-2000 CDMA Direct Spread is an air interface standard found in 3G mobile telecommunications networks. Once done, you are ready to make your UMTS/3G VoIP calls.

SIP VoIP settings
Installing SIP VoIP settings
AWCDMA

Few days ago we finally got our new optical connection. It is working like a charm. Having optical connection and Cisco router from one side and Asterisk server from the other side was a clear sign that we could/should implement QoS for voice traffic. By default, Asterisk is using port range 10000 to 20000 for RTP streams (which is adjustable in /etc/asterisk/rtp.conf) so you have several solutions how to implement Low Latency Queuing – LLQ on Cisco router. You can:

1. use Network Based Application Recognition NBAR to recognize your RTP traffic by using command ‘match protocol rtp’ within required class map;
2. create access control list which would comprehend all the traffic which is working as UDP in port range 10000 to 20000 by executing command ‘access-list 101 permit udp any range 10000 20000 any’ and then call that access control list within class map by using command ‘match access-group 101′;
3. make your Asterisk mark your SIP or RTP traffic by default.

From above listed solutions to this problem, third solution was most logical to me, although all solutions would work. Since there is recommendation to mark traffic as close to source as possible third option was very logical thing to go for. If you are using Asterisk which is working on Linux as root user (in most cases it is working with asterisk user by default) you will need to edit your sip.conf and within section [general] you will need to add following:

tos_sip=cs3
tos_audio=ef

Once done with that, save your configuration, reload asterisk and you are ready to go for Cisco configuration.

As I already mentioned, in most cases Asterisk is using user asterisk for the Asterisk daemon. If that is a case, above listed solution won’t work for you because Linux as operating system won’t allow you to mark your packages as non root user. That is very logical since we would be able to mark our web or some peer to peer traffic packages as EF (expedited forwarding) and obtain priority which is not desired at all. But, there is very powerful solution by using iptables. As root, you will need to execute following:

iptables -A OUTPUT -t mangle -p udp -m udp –dport 5060 -j DSCP –set-dscp 0×28
iptables -A OUTPUT -t mangle -p udp -m udp –sport 10000:20000 -j DSCP –set-dscp 0×28

By executing above commands you will mark your SIP and RTP traffic as DSCP CS5 (IP Precedence 5). You can verify that by catching a trace using snoop/tcpdump.

In case that you would like to set different DSCP values for some traffic, please find partial list here:
Expedited Forwarding, DSCP = 0x2E
CS5, DSCP = 0×28
CS6, DSCP = 0×30
CS7, DSCP = 0×38

Now last thing that we would need to do on the Linux side is to add those iptables rules to load on boot. Edit /etc/rc.local using your favorite text editor and paste above listed iptables lines and save it. Please note that rc.local file is loaded after the network, so if you are planning to put some firewall lines beside those that mark sip and trp traffic you might end up without your firewall rules for few seconds (otherwise, you can execute ‘iptables-save >/etc/iptables.rules‘ and then you would just need to add following line ‘pre-up iptables-restore < /etc/iptables.rules‘ to ‘/etc/network/interfaces‘ by using your favorite text editor to be loaded with network).

On the Cisco router side, we will configure Low Latency Queueing – LLQ to put some priority onto our CS5 marked packages. First thing that we need to do is to create class map from the global configuration mode by executing following commands:

router>enable
router#conf t
router(config)#class-map match-any voice
router(config-cmap)#match ip dscp cs5
router(config-cmap)#exit
router(config)#

By creating class map as listed above we would select our Asterisk marked cs5 packages. Next thing that we need to do is to define what we want to do with above selected cs5 marked packages. We are doing that by creating policy map as follows:

router(config)#policy-map llq
router(config-pmap)#class voice
router(config-pmap-c)#priority percent 20
router(config-pmap-c)#exit
router(config-pmap)#class class-default
router(config-pmap-c)#fair-queue

As per above configuration we are creating policy map called llq, and within that we are specifying 20 percent of available bandwidth for class voice. Please note that we need to have correct bandwidth statement under interface which is connected to WAN. Also, please note that we have enabled fair queueing on class-default (all data which is not selected by some other class map).

Last thing to do in regards to configuration is to enable policy map on interface which is facing connection towards your Internet Service Provider (WAN interface). We can do that by navigating to interface configuration mode and applying service policy in output direction as per following example:

router(config-pmap-c)#exit
router(config-pmap)#exit
router(config)#interface fastEthernet0/1
router(config-if)#service-policy output llq

And that is all. Now, we need to check results of our work by executing following command:

router#show policy-map interface fastEthernet 0/1
FastEthernet0/1

Service-policy output: llq

queue stats for all priority classes:
Queueing
queue limit 64 packets
(queue depth/total drops/no-buffer drops) 0/0/0
(pkts output/bytes output) 947031/193919404

Class-map: voice (match-any)
856803 packets, 170900643 bytes
1 minute offered rate 22000 bps, drop rate 0 bps
Match: ip dscp cs5 (40)
856803 packets, 170900643 bytes
1 minute rate 0 bps
Priority: 20% (400 kbps), burst bytes 15000, b/w exceed drops: 0

Class-map: class-default (match-any)
5321936 packets, 1534281066 bytes
1 minute offered rate 20000 bps, drop rate 0 bps
Match: any
Queueing
queue limit 64 packets
(queue depth/total drops/no-buffer drops/flowdrops) 0/0/0/0
(pkts output/bytes output) 5322316/1531545858
Fair-queue: per-flow queue limit 16

Please note packet counts under voice class on this policy. That means that we have deployed working configuration which will improve your voip experience. Your voip traffic will get priority among other aggressive traffic flows. Please find some additional information below, and feel free to contact me in order that you have additional questions.

1. Low Latency Queueing
2. Quality of Service

One of the biggest IT conferences, and most probably biggest GNU/Linux conference, Debconf (Debian Developers Conference) will take place in Banja Luka, Bosnia and Hercegovina in 2011! Among very professional bids from Munchen (Germany) and Quito (Equator), our Banja Luka bid won. Everything was carefully prepared and well organised by local team with Adnan Hodzic as lead person. We need to mention that we have full support which include funding, venues and else from Government of Republika Srpska which signed support letter (you can find it on Banja Luka bid wiki page). As the person which participated at Debconf7 in Edinburgh I am really happy that I will have opportunity to see known people, but also that Bosnia and Hercegovina will host this since this is our second bid (Sarajevo bid for dc7). It is our pleasure to have opportunity to host Debconf11 and we hope to see you here next year!

1. Banja Luka bid wiki page
2. My dc7 summary blog page

More than year ago when I started using Nokia Eseries one of the reasons to switch to new mobile platform was SIP stack and client support with Eseries (I used to have Nokia E71, now I am proud owner of Nokia E72). By that, I was able to connect to Asterisk or Cisco Unified Communications Manager (by using SIP digest authentication) and that was working like a charm (I am still using SIP integration heavily).

Then, I started to think about different approaches with Nokia in business environment, followed with different cost saving strategies so i asked myself about Nokia Eseries integration with Unified Communication Manager (UCCM and CME environment) by using SCCP. That was logical thing to think of, since there is SIP support already integrated into this phone series and there are plenty of Call Manager deployments in production worldwide natively working with Skinny (SCCP stands for Skinny Client Control Protocol which is often just called Skinny). And, as expected, there was really nice integration prepared by Nokia for their business users called Nokia Call Connect for Cisco.

Nokia Call Connect for Cisco integrates compatible Nokia Eseries devices with compatible enterprise voice infrastructure. When you enter the coverage area of the office wireless local area network, your device automatically registers to Cisco Unified Communications Manager and thereby activates business mode. In business mode, you can use Cisco Unified Communications Manager services to handle business calls.

With Call Connect, you can:

• Use high-speed WLANs instead of cellular networks to make calls when you are within WLAN coverage
• Use the services of Cisco Unified Communications Manager to handle business calls
• Route calls over the enterprise voice and data network to help minimize mobile phone bills
• Benefit from improved mobile phone coverage within buildings by using high-speed WLANs
• Receive notifications of new voice mail as text messages
• Access online services, such as corporate directory

Solution deployment can be splited in two parts:

• Configuring Unified Communications Manager
• Configuring Nokia Eseries phone

Since I am working in lab environment with Unified Communications Manager 6.0 and Nokia E72, I needed to make sure to download proper required packages, as follows:

Nokia Call Connect for Cisco client v2.0 (v2.0(1005)) (SIS, 1,66 MB)
NOTE: Supported devices include Nokia E52, Nokia E55, Nokia E63, Nokia E66, Nokia E71, Nokia E72 and Nokia E75

Cisco option package (COP) file for CUCM 4.x, 5.x and 6.0 for Nokia Call Connect for Cisco clients (.zip, 8 kB)
The file should be imported to Cisco Unified Communications Manager server 4.x, 5.x and 6.0 to add the Nokia Eseries devices in the device list of Communications Manager if the correct device type isn’t yet included there.

Please note that client file provided in the list above is actually trail version of the Nokia Call Connect for Cisco, and it is going to be active for 60 days after which it is going to expire and you will need to purchase real license. This text is not going to describe how to install this client, but I will suggest to install it from OVI store (it is easiest and most convenient way to do so).

Cisco Unified Communications Manager Configuration

After we have downloaded above files, we need to import Cisco Option Package to Unified Communications Manager. Once we are done with that process we will have Nokia S60 listed as valid phone type in CallManager Phones configuration. Process of adding COP file is straight forward and is described in following sections. Please note that you will need to have up and running FTP server in your network to accomplish installation of COP file.

First, we need to navigate to Cisco Unified Communications Manager Serviceability configuration pages (selection can be made from upper right corner as shown on screenshot):

Once logged into Serviceability configuration pages, we need to navigate to Software Upgrades drop down menu, and we need to select Install/Upgrade:

Once there, we need to assume control if there was some previous session:

Next thing is to select preferred source of installation. In our case that is going to be FTP server. Valid options are Remote Filesystem and CD/DVD.

Please fill all required fields (fields indicated with *). Also, make sure to put COP file in root directory of your FTP server, so that CallManager is able to find it as valid upgrade option. You will need to provide Directory (put / for root on your FTP server), Server (IP address of your FTP server), Username and Password (valid user information) and Transfer protocol which can be SFTP and FTP.

Once done with filling up required field, press Next and Call Manager will attempt to contact FTP server. If there is valid COP file (valid upgrade option), and if we are working with proper user information and running FTP server, Call Manager will list valid upgrade options  as per following screenshot:

Please note that valid COP file for Nokia S60 phone type is called cmterm-nokia_s60_001-sccp.cop.sgn. If that is what you have listed, press Next and importing process will start (downloading):

Once it is downloaded you will be presented with MD5 hash value which you can compare with one provided by Nokia from security reasons:

Once you press Next, import process will start and you will be presented with progress as follows:

Process will run for few minutes and once it has been completed, you will be prompted about it as follows:

Once done with this step, we have imported new phone type to Call Manager: Nokia S60. To make sure that it is there, we will need to check it in Cisco Unified CM Administration configuration pages. Therefore, we will need to make proper selection in upper right corner:

Once we are in Cisco Unified CM Administration pages, we need to navigate to Device drop down menu and we need to select Phone.

Once there, we will need to add new phone, and we can do that by pressing Add new button as per following picture:

Once the page is loaded we need to make proper selection, and in our case we need to select Nokia S60. If we are able to see Nokia S60 then our COP import was successful.

Now, assuming that you have installed Call Connect client (.sis) on your Nokia Eseries device and that it works fine, we can continue with configuration. In this section, we will add new phone with phone type device Nokia S60 as listed above. Please notice that product type is now listed as Nokia S60 and that Device protocol is Skinny:

Next thing that we need to do is to check  wireless MAC address on phone since it is one of the required fields in order to add new phone. Fastest way to achieve that on Nokia E71/E72 is to type following code: *#62209526# and you will be prompted with WLAN MAC address. Once you type that into MAC address field, Description field will be populated automatically. Please note that we need to fill up all the fields indicated with asterisk (*).

We will needed to select Phone Button Template and Commong Phone Profile fields since they are required. Also, that includes Presence Groups and Device Security Profile fields as shown on following screenshot:

Once done with basic configuration, we need to save changes by hiting Save button. Then, we need to configure associated information which includes configuration of directory numbers as per following:

Once there, we need to click on Line [1] – Add a new DN and we need to fill up required fields. Field of interest is Directory number. Also, please notice that in Associated Devices box, our Nokia Eseries device will be listed:

Once done with directory number configuration, click on Save and you will have your new Nokia S60 phone configured and listed on phone list, as follows:

Once we are done with configuring Cisco Unified Communications Manager side which includes importing COP file and configuring new phone, we can start configuring Call Connect client which we installed on Nokia Eseries phone.

Configuring Nokia Eseries phone

Since we have server side up and running, we can start configuring our Nokia Eseries Call Connect client. In our lab environment, we are using Nokia E72 and following screenshots are taken on that phone. First thing that we need to do is to open installed application by navigating to Menu > Apps > Nokia CC Cisco. Please notice that Call Connect is offering multiple productivity features such as Call pick up, Group Call pickup, Call divert and DND. Also, please notice that we do not have SCCP active profiles, and in following sections we will describe process of configuring one.

To configure new SCCP service, select Options > Settings > New profile

Once in the New profile configuration mode, we will need to configure Profile name, select default Access Point and configure TFTP server. Please note that in our example Profile name is set to Call Manager, 6BFlat5 is default access point and that TFTP is manually set to 192.168.1.10 which is in our case IP address of our lab Call Manager. Valid option for selecting TFTP server is also DHCP, but in that case we would need to configure DHCP server with option 150, which would indicate IP address of our TFTP server.

Once we are done with basic profile configuration, we can click on Back. We will see our new profile in “Not registered” state. To register service, we will need to navigate to Contacts and then from viable options drop down list we need to select Cisco VoIP and select “Activate service”, as shown below:

Once you activate the service, you will see your newly created profile registered.

Once registered, one additional step can be made in order to make sure that all is working properly. Navigate to Menu > Apps > Nokia CC Cisco and select Status information. You will be able to check what is the Stack version, Outgoing phone number, License information, MAC address, DHCP related information, Networking information and SCCP profile information.

To make sure that all is running fine on Unified Communications Manager, navigate to Cisco Unified CM Administration configuration pages, select Devices drop down list and from there pick up Phones and click on Find/List. You should receive output that indicates that SCCP phone is registered, as follows:

Please notice that in upper right corner on your phone you will be able to see your configured directory number followed by the SCCP profile name (in our example it is (1003)CallManager), and also, registration status will be indicated by the small VoIP icon in bottom part of the screen of your Nokia Eseries phone. Once you have your profile registered with Unified Communications Manager, you can start making VoIP phone calls and you can start using productivity features that we already mentioned in previous text.

For more details about Call Connect please refer to following links:

1. Nokia Call Connect For Cisco
2. Nokia Call Connect For Cisco: Licensing and Support

After a long time, Zenica GNU/Linux Users are about to have meeting! It is going to take a place in Caffe “Ekran” at 18h on 31. Jan ’10. We intend to talk about some future projects, Linux based solutions that we have learned about during our many months of inactivity, and we intend to have good time. Since we were inactive for some time we are happy that there are some new users that we intend to meet with on this meeting. If you are interested in our projects and who we are, just show up. We will be willing to meet you, help you with your Linux/Networking/Programing/IT issues (you don’t have to be Linux user as you can see, and we can help up to level that we are capable to help) and to share our experience!

1. Linux Users group of Bosnia and Hercegovina
2. Zenica Linux users forum

If you are using some Cisco router which is capable of modular QoS you are able to get statistics about bandwidth consumption by specific protocol in your network. Cisco modular QoS is using Network Based Application Recognizition, feature that you will like a lot if you are working with QoS and feature that will make your networking job lot easier. Basically, NBAR is able to recognize type of application/protocol which is communicating over network, and based on that you are able to manipulate that data. You could mark that traffic, shape or police it. This blog post won’t cover those techniques, but my intention is to show how to get statistics. Procedure is very simple:

1. Verify your interface configuration:

R1#show ip interface brief
Interface                  IP-Address      OK? Method Status                Protocol
FastEthernet0/0            80.239.11.100   YES NVRAM  up                    up
FastEthernet0/1            unassigned      YES NVRAM  up                    up
FastEthernet0/1.1          192.168.1.1     YES NVRAM  up                    up
FastEthernet0/1.20         192.168.20.1    YES NVRAM  up                    up
FastEthernet0/1.40         192.168.40.1    YES NVRAM  up                    up
NVI0                       unassigned      NO  unset  up                    up
Tunnel0                    10.100.100.1      YES NVRAM  up                    up

Verify which interface your WAN is working on. In this case it is FastEthernet0/0. We will use that interface for our statistics.

2. Navigate to global configuration mode, and then to interface configuration mode:

R1#
R1#conf t
Enter configuration commands, one per line.  End with CNTL/Z.
R1(config)#interface FastEthernet 0/0

3. While in interface configuration mode, activate NBAR protocol discovery:

R1(config-if)#ip nbar protocol-discovery
R1(config-if)#

4. It would be good to tune load interval for statistics that we will gather from default 5 minutes to 1 minute:

R1(config-if)#load-interval 60
R1(config-if)#

5. Next, we need to issue proper show command to get statistics, and that would be:

R1#show ip nbar protocol-discovery stats bit-rate top-n 10

FastEthernet0/0
Input                    Output
—–                    ——
Protocol                 1min Bit Rate (bps)      1min Bit Rate (bps)
———————— ———————— ————————
dhcp                     22000                    0
http                     0                        1000
gre                      0                        0
rtp                      0                        0
ipsec                    0                        0
secure-http              0                        0
ssh                      0                        0
dns                      0                        0
icmp                     0                        0
snmp                     0                        0
unknown                  64000                    0
Total                    86000                    1000

As you can see in above example, we are able to get statistics by protocol on specific interface in 1 min bit rate in inbound and outbound direction. Based on those statistics you can make some decisions, what needs to be blocked, shaped, policed or marked. I found this to be a first logical step when deploying QoS.

6. Optional step would be to create alias for show command that is being used in above example:

alias exec traffic ip nbar protocol-discovery stats bit-rate top-n 10

Now, once we type traffic command in privileged mode, we will get protocol statistics:

R1#traffic

FastEthernet0/0
Input                    Output
—–                    ——
Protocol                 1min Bit Rate (bps)      1min Bit Rate (bps)
———————— ———————— ————————
dhcp                     22000                    0
http                     0                        1000
gre                      0                        0
rtp                      0                        0
ipsec                    0                        0
secure-http              0                        0
ssh                      0                        0
dns                      0                        0
icmp                     0                        0
snmp                     0                        0
unknown                  64000                    0
Total                    86000                    1000

For more information please refer to following links:

1. Modular QoS
2. Quality of Service

During last few months I was intensively working with Cisco Unified Communications Manager, previously called just Call Manager in order to obtain Cisco Unified Comminications IP Telephony (CIPT) certification. Since I am coming from the ‘voice’ field when I started to work with Cisco products it was logical to me to check what is Cisco offering in that field. And I remember that I was impressed. Six different certifications after CCNA and two possible CCVP paths. Plenty of different solutions, gateways, protocols and such was enough challenging to me. First thing that I needed is to make clear decision of which CCVP path to follow. One is covering CallManager (version 6.X called CUCM) in two parts (CIPT1 and CIPT2) and other, old one, is covering CallManager (Cisco Unified CallManager 4.X) throught one certification mixing everything with additional Gateway/Gatekeeper certification and that path is about to reach end of life on December, 31. Since I was working with web based call processing device in past I decided to go with actual Unified Communications Manager CCVP path (because CUCM is web based call processing solution as well). One of the exams on that path is Cisco Unified Communications IP Telephony Part 1 which I have passed today.  There was 60 questions and you needed to score around 80% to pass it. There are single choice, multiple choice and drag and drop questions. It was not that easy at all although I was preparing for it for few months and that is normal because this is very complex solution. When it comes to CUCM I need to say that I was quite surprised of number of features that it can provide. It is very powerful, high available and redundant call processing solution which is covering advanced mobility, call coverage and other solutions in very organized, logical and intelligent way. I was preparing my certification following multiple documentation sources and by following quick reference. Note that CallManager can be installed in VMware which is a good thing, because when it comes to practicing you won’t spend lot of money to build complete testing environment. Also, please note that there are many good CCVP blogs which can help a lot and I will post few links below the text. If you need some additional information on CallManager or this certification, feel free to contact me, I will be willing to help.

1. Cisco Unified Communications Manager
2. CCVP certification paths
3. CCIE12932 blog
4. Chris’ CCVP blog

One of the best Cisco Unified Communications Manager VoIP features is Extension Mobility in my personal opinion. It allows you to temporarily configure another IP Phone as your own by logging in to that phone. Once logged in you will have your number, speed dials and etc. onto that phone, and if you are working as teleworker you would know to appreciate those kind of options. Following text describes how to configure CallManager 6.0 to support Extension Mobility.

Task 1: Verify Extension Mobility Service is Running

Step 1: From the Navigation menu select Cisco Unified CallManager Serviceability

Step 2: Select Tools>Control Center – Feature Services

Step 3: Make sure that the Cisco Extension Mobility service shows status Activated

Task 2: Configure Extension Mobility Service

Step 1: From the Navigation menu select Cisco Unified CallManager Administration

Step 2: Select Device>Device Settings>Phone Services

Step 3: Click Add New

Step 4: In the Service Name field, type Extension Mobility
Step 5: In the Service Description field, type Login and logout service
Step 6: In the Service URL field, Enter the following URL: http://YOURCUCMIPADDRESS/emapp/EMAppServlet?device=#DEVICENAME#

Step 7: Click Save

Task 3: Modify Enterprise Parameters to Reflect IP Address of CallManager (remove DNS reliance)

Step 1: Select System>Enterprise Parameters

Step 2: Under Phone URL parameters, change all fields to reflect IP addresses instead of hostnames. Change ONLY the host name, not the reset of the field.

Step 3: Click Save
Step 4: Click Ok from the pop-up warning.
Step 5: Click Reset
Step 6: In the pop-up window select Reset
Step 7: Click Close

Task 4: Create Device Profile Default for Each Phone Model that shall Support Cisco Extension Mobility (this step is optional)

Step 1: Select Device>Device Settings>Default Device Profile
Step 2: From the drop down list, select the phone model to be configured, for example, Cisco 7960.
Step 3: Under Description, enter a description of this profile.
Step 4: Under Phone Button Template, select Standard 7960 SCCP.
Step 5: Click Save
Step 6: Repeat for each model phone to be configured

Task 5: Create Device User Profile for a User

Step 1:  Choose Device>Device Settings>Device Profile and click Add New.

Step 2: From the drop down list, select the phone model to be configured, for example, Cisco 7960
Step 3: Click Next
Step 4: Enter a Device Profile Name (in this example KemalSanjtaProfile).
Step 5: From the Phone Button Template field, select Standard 7960 SCCP.
Step 6: Click Save.

Step 7: On the left hand side of the screen, click the link Line [1] – Add a new DN.

Step 8: Choose a valid DN from your NIP, enter that DN in the Directory Number field.
Step 9: Under Route Partition, select your city’s Headquarters Partition.

Step 10: Under Directory Number Settings choose a CSS of appropriate access.

Step 11:  Enter any Call Forward and Call Pickup Settings as necessary.
Step 12: In the Display (Internal Caller ID)
Step 13: Click Save.
Step 14: From the Related Links: menu, select Subscribe/Unsubscribe Services.

Step 15: In the Select a Service, select Extension Mobility, then click Next.

Step 16: Click Subscribe.

Step 17: Click Save.
Step 18: Repeat steps 7-13 for any additional lines.

Task 6: Associate User Device Profile to a User

Step 1: From the menu, select User Management>End User.

Step 2: Click Find
Step 3: Select the user from the list that matches the profile that was created.

Step 4: Under Extension Mobility>Available Profiles, select the profile that was created in the previous exercise and move it to the Controlled Profiles selection (in our example it is KemalSanjtaProfile).

Step 5: Under Default Profile, select the profile.
Step 6: Click Save.

Task 7: Configure and Subscribe Cisco Unified Ip Phones to Service and Enable it.

Step 1:  Select Device>Phone from the menu.

Step 2:  Select the phone from the list of devices.

Step 3: In the Related Links: field, select Subscribe/Unsubscribe Services and click Go

Step 4: In the pop-up window, under Service Information, in the Select a Service pull down menu, select Extension Mobility.

Step 5: Click Next
Step 6: Click Subscribe

Step 7: Click Save

Step 8: Close the pop-up window

Step 9: Under Extension Information , check the Enable Extension Mobility box.
Step 10: Under the Logout Profile field, select – Use Current Device Settings –
Step 11: Click Save.

Step 12: Click Ok from the pop-up warning.
Step 13: Click Reset
Step 14: In the pop-up window select Reset.
Step 15: Click Close.

Note: This post has been updated on 12/03/2010 in order to describe how to assign Extension Mobility Phone service to Device Profile (including screenshots).