WhichVoIP Blog

The previous blog explained why we have 8 bits in a DS0, here we’ll explain how we get these 8 bits.

When  your voice is digitally sampled the amplitude or how loud you are speaking is captured, this happens 8000 times a second. However this digital sample is not what is sent. For a T1 line an algorithm called u-law is used to compand the data. This is a logarithmic type of algorithm that has the effect of giving greater resolution to lower value samples i.e. when you are talking quietly, and less resolution to higher amplitude samples. This intuitively makes sense, if someone is shouting then you can afford to loose a little differentiation between sounds but when someone is whispering you want to hear or accentuate the different levels so you can make out what they say. Typically when your voice is sampled, an Analog to Digital converter that has a 13 bit output is used, the u-law algorithm as well as performing this logarithmic compression has the effect of reducing the voice sample to the 8 bits that is sent.

Up until now we have confined our discussions to voice T1’s. Next time we’ll take a look at how data is transmitted via a T1 line.

WhichVoIP team

http://www.whichvoip.com

How did the 8KHz frame rate and the 8 bits per DS0 come about?

I’ll take the easy one first - Why 8 bits per DS0? For digital or telecom engineers 8 or any multiple of 2 is a perfect number. It’s very easy in digital logic to count in multiples of two thus it is very easy to extract these DS0’s, generate the frequencies and clocks required to receive and transmit at these rates and multiplex these T1 streams into T3 streams or higher rates. The trickier question is how do we get our voice to be transferred from one point to another in only 8 bits every 125us(this is the equivalent time of an 8KHz frame rate).

There is a theorem called Shannons theorem or sometimes referred to as Nyquist-Shannons theorem that drives the fundametal basics of all digital communications.  First remember that T1’s were invented long before the advent of the internet or VoIP and they were designed with the specific goal of transferring voice in a digital format. The result of Shannons theorem is that to sample an analog signal(your voice in this case), convert it to a digital format and then reconstitute it as an analog signal(your voice at  the other end) that has lost no or little information(the person you are talking to can understand you) the original analog signal must be sampled at a rate of at least twice the highest frequency you want to transmit.  The typical human voice has a frequency range of about 50Hz to 3500Hz. So if you want to sample that voice and digitize it you have to sample it at a rate of at least 7000Hz. As in all good designs there has to be some design margin, often referred to as a guard band in communication theory, a rate of 8000Hz was chosen. The frame rate of 8KHz became the standard.

Now we know why 8 bits is used in a DS0 and the rate it must be sent at. In the next blog we’ll go into a little more detail on how the 8 bits are generated and what they represent.

WhichVoIP Team

http://www.whichvoip.com

There has been a lot of publicity recently on Asterisk for your business telephony needs. 

What is Asterisk?

Asterisk is known as the open source IP PBX.  Basically, rather than using traditional PBX equipment or Cisco VoIP platform equipment, such as their Call Manager IP PBX, some businesses and universities are utilizing Linux based servers to handle their telephone calls using VoIP.  No expensive PBX equipment needed and no Cisco license fees as part of their call manager network.

The linux servers are normally found in such environments anyway in order to handle the data network for a facility so adding voice is a case of installing asterisk and connecting your phones to the server.  Analog phones will need some form of conversion to digital (e.g. using Cisco VGC gateway devices), but if you have IP phones, simple, just connect the Ethernet cables from the phone to the server.  The voice traffic will simply follow the same path as the data does now over the T1 interfaces. 

There is some downside of course.  The technical support has gone, although so has the expense   So long as you are comfortable running your voice and data network with Linux servers and have some expertise in this area so you can handle the administration, the cost savings are substantial.  Recent University studies suggest that moving from a Cisco call manager license based system to Asterisk can reduce your telephone costs by over 65%.

Asterisk is definitely one for us all to watch.
http://www.asterisk.org/

Andy

If you have ever searched for Business VoIP or have been serious about using any business telecom service then you will have come across the term T1. Over the course of the next few weeks we’ll endeavour to explain what a T1 is and delve down into the details of it for you techies out there.

So what is a T1? Well a T1 really defines 4 things.

1. The physical medium used.
2. The bandwidth or clock rate.
3. The type of signaling and encoding used.
4. The format or frame the data is transferred in. 

With a T1 there are two physical mediums. One is 100 Ohm shielded twisted pair and the other is 75 Ohm Coaxial cable. Typically inside patch panels the 100 Ohm twisted pair is used as it is cheap and flexible but for transmission to the outside world a balun is used to match to 75 Ohm Coaxial cable which is typically used to connect to the ISP or PBX.

The bandwidth of a T1 is 1.544Mbits/sec that’s for one direction, one of the advantages of a T1 is that you get this dedicated bandwidth in both the uplink and downlink directions.

T1 uses a signal encoding called B8ZS which stands for Binary 8-zero Substitution - quite a mouthful. Without going into too much detail just now, this type of encoding allows data and a clock to be sent across the same cable - vital for synchronous data transfer.

Finally the format - Each T1 is split into 24 channels referred to DS0’s. Each channel is 8 bits wide and the frame is sent on a 8KHz boundary. The data capacity is therefore 24×8x8000 = 1.536Mbits/sec. Didn’t I say earlier that a T1 was 1.544Mbits/sec? Well there is something called a framing bit, this is 1 bit every frame that a receiver on the other end can lock onto to ensure that whatever equipment is there can extract the correct DS0’s in the correct order. This framing bit makes for an extra 8000 bits/sec giving a grand total of 1.544Mbits/sec.

T1’s are essential a for any business that employs more than about ten people at the one site. They are very reliable and give you a guaranteed bandwidth.

If you are starting a business or you already own a  business and are planning on expanding, fill out this form for a quote on how much a T1 is - you maybe pleasantly surprised.

Next week or when we have time we’ll explain how the 8KHz frame rate came about and how we get 8 bits in a DS0.

WhichVoIP team

http://www.whichvoip.com

Have you ever walked into work on a Monday morning and found this fancy new digital phone on your desk? Well if you have (and the odds are increasing fast), then I am guessing that it is unlikely that you stopped for a second to think about what was actually involved in getting it there. Hopefully you have not begun your Business VoIP experience by cursing the phone because you are having trouble using all the features. As such I hope you received proper training as to how to use the nifty new features that a Business VoIP system can provide.

Having installed a bunch of these systems in various sized organizations across the country, I thought that someone may be interested in the high level process of what is involved in transitioning a business from a legacy analog phone system to a new VoIP system. Who the heck does these installations anyway? Well it is extremely unlikely that a Cisco or an Avaya will actually do the installation for you. These companies will fight tooth and nail to convince you to purchase its voip solution, but after that, there is very little involvement from them. Unless of course you are part of a large enterprise which can demand and receive direct support.

So in the majority of cases a partner (e.g.   ) of these companies will actually perform the installation and manage the entire VoIP project. These partners are more commonly known as VARs or Value Added Resellers and are pretty much made up of a bunch a sales guys and a group of engineers specializing in various technologies such as telephony, security, wireless, etc. It is these engineers that actually do the installation, with the VAR charging the end customer anywhere between $150 and $250 an hour for its services. Now that has got your attention, hasn’t it? Yes, there is a lot of money to be made through VoIP. If only the poor engineer got paid that much. Well actually, a good VoIP engineer who can design and deploy an entire Business VoIP solution, will typically earn well over $100K per year in most parts of the country.

Next week I’ll go into some more detail on the actual process that companies go through when swapping over to Business VoIP .

Tony

I really enjoy reading the many VoIP reviews we receive each day at WhichVoIP.  The majority of these VoIP reviews are very informative, carefully written and extremely helpful to other readers trying to decide which VoIP service provider is best for them.  We get many promoting a particular VoIP provider, some critical of them and a few that are just plain ugly

Our feedback form for user postings really does attempt to get a true reflection of the VoIP provider in question.  We ask for feedback on such things as voice quality, reliability, customer service, time to install and many other important factors.  The benefit of this is of course that our current readers interested in selecting a VoIP service provider can collect a vast amount of data to assist in the decision making process.

A word of caution however, often those who have had a bad VoIP experience are the ones most willing to post a review, which of course turns out to be negative!  We do receive a lot of very good VoIP reviews too but those happy with their service have little reason to be looking at our site and post a user review - why would they be shopping for another VoIP service provider?

What is our current best reviewed VoIP service provider you ask?  Well, if we disregard those VoIP providers that have less than 5 user review postings, the current winner is , with an overall score of 4.7 out of 5 from 16 postings.

Our VoIP review section is updated daily, take a peek and read some very informative user reviews….some will give you a laugh too. 

Andy