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.
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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.
While reading the tedious bandwidth consumption chapter and how to calculate it, I found that between the endless numbers there is a small light. I do not know if the exam expect me to calculate bandwidth (and I do hope it does not) but for real life there are few options that make your life easier.
The first option is Cisco’s Voice Codec Bandwidth Calculator which is only available if you have a valid CCO login. This is how it would look (using totally random options):
This is where you choose the Codec and the Voice protocol to be used.
In addition you have to type in the number of calls you expect on the line – this is for simultaneous calls.
Here you have to type your Voice Payload in bytes.
The other parameters are the Media access, this is the type of connection you use and the Tunnel or Security load (in bytes) that would be used.
When you submit your information you will get a detailed output with your selections and results. This report is full of numbers which I guess will clear during my studies but even now, when I’m short of some definitions I can appreciate the information and it is obviously very helpful in planning a voice network
This is an example of a result page:
and the second part of the results:
This is a very details tool but as I mentioned, it requires a valid CCO and not everyone can pay for it (though I assume that if you have this type of voice network you can and should get maintenance for your equipment)
The second option is a free four Voice over IP calculators. While it is not as descriptive and comprehensive as the Cisco version, it does give you the general idea with good numbers. Check this example:
Unlike the fancy Cisco tool, this free tool have fewer options of coding algorithms.
After you select the packet duration (samples over time) you can either type in the number of simultaneous calls and receive the required bandwidth OR type the available bandwidth and receive the number of possible calls.
The free version has three more options:
Erlangs and Bandwidth Calculator
Minutes and Lines Calculator
Erlangs and Lines Calculator
One more tool from Cisco (available for CCO only) is DSP Calculator which take the router model and IOS version, on board HWIC slots and few more parameters and give the DSP Module Requirements. I think it is beyond the scope of the exam but it is a handy tool.
Last night I started watching the introduction to VoIP. While the ‘why’ is easy to understand and been part of my work for years, the ‘how’ is new and unfamiliar.
The review of different devices and topologies wasn’t exciting, it is after all a basic introduction but one piece caught me and sent me digging: the Nyquist Theorem
Harry Nyquist is the son of Lars Jonsson, he was the fourth child of eight and was born on February 7th 1889 in Nilsby, Sweden.
I guess this is where the story get interesting – the name Jonsson had to be changed because about a hundred meters away from their house lived another Lars Jonsson and there was huge problem with the mail delivery…
Harry’s father agreed to change names (which wasn’t a rare thing to do at the time), he changed the name to Nyquist. Isn’t it amazing?
I wonder what was Harry’s dad thinking when he picked this weird name…
When Harry moved to the USA he was able to get great education and achieved his Ph.D. in physics in 1917 at Yale. nice!
His working career started working for AT&T and then Bell from 1917 for 37 years he had a big part in many inventions including the FAX and more than 100 patents.
Read more on Harry Nyquist’s biography here.
One of his big achievements which is also the reason we’re here is the Nyquist Theorem. A great achievements that even today, with all the technological progress is a foundation to voice based technologies.
The Nyquist theorem states that a signal must be sampled at least twice as fast as the bandwidth of the signal to accurately reconstruct the waveform; otherwise, the high-frequency content will alias at a frequency inside the spectrum of interest (passband). An alias is a false lower frequency component that appears in sampled data acquired at too low a sampling rate.
- Pulse Code Modulation – The first step to convert the signal from analog to digital is to filter out the higher frequency component of the signal. Most of the energy of spoken language is somewhere between 200 or 300 hertz and about 2700 or 2800 hertz.
The second step to convert an analog voice signal to a digital voice signal is to sample the Filtered input signal at a constant sampling frequency. The pulse train moves at a constant frequency, called the sampling frequency. The analog voice signal can be sampled at a million times per second or at two to three times per second.
the next step is to digitize these samples in preparation for transmission over a Telephony network. The process of digitizing analog voice signals is called PCM.
- Differential Pulse Code Modulation- This is a similar but improved technique to cut the size of the data for the same content.
Differential PCM (DPCM) is designed to calculate this difference and then transmit this small difference signal instead of the entire input sample signal. Since the difference between input samples is less than an entire input sample, the number of bits required for transmission is reduced. Using DPCM can reduce the bit rate of voice transmission down to 48 kbps.
My favorite part is the revelation of 64:
As the input signal samples enter the quantization phase, they are assigned to a quantization interval. All quantization intervals are equally spaced (uniform quantization) throughout the dynamic range of the input analog signal. Each quantization interval is assigned a discrete value in the form of a binary code word. The standard word size used is eight bits. If an input analog signal is sampled 8000 times per second and each sample is given a code word that is eight bits long, then the maximum transmission bit rate for Telephony systems using PCM is 64,000 bits per second. In other words, if we sample the sound frequent enough we’ll have enough data to transmit and rebuild it on the other side. Nyquist suggested double or more and the reason is that the edges of the wave change faster and will not be able to reconstruct unless we get at least twice the samples.
If you’re into the numbers behind the theory (or shall I say if you’re a real geek) This Document have the formulas and details that are way beyond the exam scope or my interest.
Today I’m doing the first step toward the last piece of my CCNP puzzle, ONT studies. Unlike the other three exams which included materials I’m familiar with from work, ONT’s Voice and QoS concepts are a new territory. My work environment include Avaya equipment (is it okay to mention this name on a Cisco post?) but other than some basic tasks I do not manage it so that does not count.
My plan is watching Jeremy Cioara’s CBT Nuggets video and hopefully by the time I complete the series it will have a different feeling. I trust Jeremy’s enthusiasm to do the job as he did so well before.
Only 138 left to complete my CCNP (but I think I won’t need them ;))
Deploying Cisco Wide Area Application Services from CiscoPress is not for everyone.
This is a concept book that aim at professionals involved with design and deployment of WAAS in their networks.
What is WAAS?
In simple words it is a solution (combination of design and technology) to get better performance out of your applications over WAN connections.
The problem (again, in simple words) is closing the gap between the need for bandwidth ($$$) and performance. Some (but not all) of the problem is getting more out of the existing infrastructure. The other part is technology related issues, mostly with the way our cloud is designed.
The book is a great read with detailed descriptions and it covers many design options. It is worth reading if only for the concepts.
After a short introduction the authors review the requirements, planning and implementation.
They cover different scenarios and many options.
They cover different type of connections (like Branch office or Data center) and go over the configuration.
This book has all the steps of the solution and you can either review it all or check out the one piece you’re interested in.
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