Quality of Service. Three words that describe (almost) a full exam.
When normal person hear these words they think about the supermarket cashier or the drugstore pharmacy. Some people might think of their last call to their utility company, where the automatic message announced that “this call is monitored for quality of service purposes”.
If you read this blog regularly (and you’re not my wifi) you are not a normal person. You are twisted, think only about networks and you understand that there is no such thing as quality of service, it is called QoS
What is it good for? There are few problems that QoS try to attend:
- Lack of bandwidth - QoS cannot help when there is no bandwidth left
- Packet loss - Voice packet loss affect the quality of call. While data transfer (Internet or FTP) is hardly affected and the user will not notice small hiccups, voice users will notice immediately.
- Delay - Same issue as in packet loss. A regular data usage will not be notice small delays while voice\video are heavily affected.
- Jitter = Delay Variation – Variable form of delay. A difference from when a packet is sent to the next packet => overall delay
- Classification – Identify and group different traffic types. Not critical apps and Important apps. Matching the different types of applications.
MATCHing is done using ACL – this is very processor intensive.
- Marking – Taging the packet so it can be quickly recognized elsewhere on the network. Marking put a tag in the header so other routers can process it faster (and save the local processing resources)
- FIFO = First In First Out – Whoever came first will be forwarded, when the buffer is full it will drop the rest of the traffic without looking at the data.
- Random Early Detection = RED – When the buffer is close to full the router can start freeing space and drop packets out of the queue.
- Weighted Random Early Detection = WRED – Cisco proprietary, allow the router to aim at the traffic it drops. This is RED with some brain.
The following will be covered in future posts so I’ll mention them but will not detail:
- Policing -> drop or mark packets when the limit is reached
- Shaping -> queue packets when the limit is reached, not dropping it
- Queuing -> method to priorities packets
Are you a Windows 64-bit user?
Are you also a BitDefender user?
If you answer YES to both questions you had a rough weekend realizing your antivirus software think your OS is a trojan…
Due to a recent update for Windows 64-bit systems it is possible that BitDefender detects several Windows and BitDefender files as infected with Trojan.FakeAlert.5
BitDefender released a fix for this. Only time will tell how their (good) reputation will be affected by this fiasco.
When I start working on a new exam one of the first things I do is checking the contents of the books. My ONT Official Exam Certification Guide have three parts: VoIP, QoS and Wireless. Out of 10 chapters Six (6) go for QoS and three (3) for wireless which leave one small chapter for VoIP.
VoIP is why we’re here to begin with. It is the reason QoS is so important and understanding it is important. This post will go over the foundations of VoIP as reflected on the ONT exam. It does not cover the in-depth of things as ONT take a relatively brief look at this topic.
First, some of the main reasons to migrate a telephony system to IP-based systems:
- More efficient use of bandwidth and equipment
- Consolidated network expenses
- Improved employee productivity
- Access to new communication devices
Personally, I think the mobility of devices in an IP network is the single most important reason to convert a network.
There are two major steps in a network migration to VoIP environment:
Phase I migration – keep existing system and connect the PBX to the router. The router can convert the calls to VoIP and save cost, QoS with low-cost.
Phase II migration – IP systems replacing all phones and PBX. This is a full IP-based environment.
Call control = routing voice around the network.
Distribution= every device has brain, every router have to be configured with all details.
Centralized = call agent is the center point with database for all calls in the network.
Cisco Call Manager (CCM) is a centralized solution.
FXS (Foreign Exchange Connections) is an analog interfaces that connect with old devices. Each analog port can run one call only, not efficient and costly. FXS ports plug to station and generate dial tone.
FXO (Foreign Exchange Office) convert analog to VoIP. It receives the dial tone.
E&M (Ear and Mouth) Receive and Transmit, create direct trunk between PBXs or between PBX to a router.
FXS, FXO and E&M are all analog, they all use one call per line.
The next step of understanding VoIP is understanding how does voice become a packet. There is a four step recipe to turn voice into bits:
- Sampling – take many samples of the analog signal
- Quantization – calculate a number representing each sample
- Encoding – convert that number to binary
- Compression (optional) – compress the signal
I wrote about waveform, Nyquist Theorem – the man and the theory so I won’t repeat myself. This is the most important part, the basic required to master QoS – my next topic that will get most of my attention.
This book is waiting on my desk for a while now. I’ve been busy with BSCI and only now found the time to open it, while working on my ONT exam.
I do some maintenance and configurations on my partial IP phone system (Avaya PBX) but it is definitely not the core of my work, not to a level that can be considered as good experience. So I’m new to the VoIP and QoS world and reviewing this book while watching the ONT videos and reading the books is very interesting.
SIP Trunking by CiscoPress is a high level resource that cover a relatively new trend in the IP telephony world.
ISPs and LAN environment converted to VoIP and IP telephony but they still use TDM trunks, the old method or in its other name – the bottleneck.
With SIP (Session Initiation Protocol) you can gain
Point-to-Point VoIP and get better service, QoS and flexibility.
The Book has a clear and very interesting introduction part. As I said, I’m not a big voice expert but I had no problem understanding the concepts, the benefits and the problems.
The second part focus on the design, planning the network for SIP trunking. It covers the component, the trunking models and design considerations. I was able to find many familiar scenarios of different types of offices and get the idea of the added value of SIP trunking.
The third and last part is the deployment of SIP trunks, this was just over my head with commands and case studies that go beyond my understanding (at least at this point of my VoIP life).
Overall I enjoyed reading this book and learned many concepts and models beyond SIP trunking. If I had to work on a SIP project this book would be a good resource and is very recommended. It is also a great book for anyone who work with telephony (IP or not) – this is where the world is moving to and even if you do not see the immediate value, it will prove beneficial on the long run.
Since I’m running a small shop I do everything and this is the type of book that allow me to better understand the big picture. I will definitely read it again after my ONT exam and I assume that with my additional knowledge and some (very basic) lab experience I will be able to get more out of it.
Check all my reviews here
It is one of those daily tasks that we don’t think about too much: renaming a computer.
Using the first initial last name method I set a machine name for my XP users. When possible I prefer the window based method as described in KB295017:
- Click Start, right-click My Computer, and then click Properties.
- Click Start, click Run, type sysdm.cpl, and then click OK.
- Click Start, click Control Panel, double-click Performance and Maintenance, and then click System.
And the next step:
- Click the Computer Name tab, and then click Change.
- Type the new computer name in the Computer name dialog box.
- Type the new domain or workgroup in either the Domain dialog box or the Workgroup dialog box.
- Click More to change the primary Domain Name System (DNS) suffix.
- Click OK three times, and then restart the computer.
This is all nice and easy to follow BUT it does not always work.
Today when I tried to rename a PC the following error prompted:
A connection to the server could not be performed because the maximum number of simultaneous connections has been reached.
It is not the first time I’ve seen it and though you would expect it to disappear after you boot the machine, the same error show up every time.
At this point there are two types of administrators: the first type is the warrior who spend hours or days to find the reason, post the problem on every forum and maybe (but not always) find a solution.
I’m the other type, the busy administrator who cannot afford spending so much time on something that can be resolved in few minutes using a different method.
This is the syntax used to change a machine name using Netdom.exe
netdom renamecomputer machine /newname:new_computername /userd:domainname\administrator_id /passwordd:* /usero:local_admin
/passwordo:* /reboot:seconds before automatic reboot
You can find the full details either on the KB page or via command line HELP.
After hitting the Enter key you’ll be prompted for one or two passwords, based on the options you choose. Few seconds later the process is completed and after you boot the computer (either using the /reboot option or manually if you didn’t use it) the machine will have a new name in Active Directory.