Subnets

When I'm teaching people about networks, the one thing that tends to cause confusion is the concept of Subnets.

If you've ever manually entered an IP address, you'll be familiar with the following:

I've added the red box to point out the "subnet mask"; a bunch of 255's and 0's that magically appear whenever you type in an IP address.

But what does it mean?

Splitting Networks

I briefly touched on the subject of packet switching in my "Layers of the Internet" article. Basically, an Internet Protocol (IP) network is a collection of devices connected via some form of network.

But what if we wanted to have some computers on the same network, but we didn't want them to interact? For example, we had the computers that controlled a company's payroll on the same network as the company's email servers.

Obviously that's not a wise thing; anyone with a bit of networking knowledge and access to Google would be able to give themselves an unexpected pay rise.

Thankfully, internet engineers foresaw this problem and came up with "subnets". A Subnet is a simple way of splitting one physical network in to two "virtual" networks.
Just like devices that are on separate physical networks, devices on different Subnets can't communicate with each other.

In other words, you can have the finance guys  on one subnet and the rest of the company on the other subnet, and you'll never have to worry about people editing their own pay grade.

One Number; two addresses

The great thing about Subnets is that they are contained within the devices' IP addresses by default.

The IP address is made up of two parts; the "Subnet" address and the "Device" address.

In fact, an IP address is a lot like a Street address. If you look at an address like "31 George St" you can instantly find that building. "31" by itself means nothing, and "George St" is too vague to be of any use.

In our IP addresses the "Subnet" address is the street, and the "Device" address is the number.

But let's look at one of the IP addresses above.

192.168.100.5

That doesn't look much like "31 George St". Which part is the Street, and which is the number?

Subnet Masks

Now we are finally coming to the mystery behind the Subnet Mask.

A Subnet mask tells us which part of the address is the street and which is the number.

The network in the diagram above has a subnet mask of:

255.255.255.0

Quite quickly we can see something special here. Everywhere that there is a "255" is part of the Street Name, and every where that there is a "0" is the number.

Let's look at that address again:

192.168.100.5

If use use the above rule, the "Street" (Subnet) becomes 192.168.100 and the Device number is 5.

Looking at our network again, we can see that the Finance machines on subnet 192.168.100 are effectively on a separate network from the workstations on the 192.168.101 subnet.

But what if...

Okay, for an exercise, let's take the same network as above, but let's change the subnet mask to:

255.255.0.0

As you'd imagine, all of the devices are now on the same subnet:

192.168.100.5

is on the same subnet as 

192.168.101.5

So we have to be careful when designing our networks to include the correct subnet masks in all of our addresses.

But what does 255.255.255.0 mean?

I have never been asked this question, so I am going to pre-empt you all and give you the answer.

Remember that all computers work in binary, that is, 0's and 1's.

An IP address is made up of four groups of eight bits, that is, eight 0's or 1's.

So, if we were to look at 192.168.100.5 in binary, it would look more like:

11000000.10101000.0110010.00000101

Now, that doesn't look like much to a normal person, but now let's have a look at the subnet mask 255.255.255.0:

11111111.11111111.11111111.00000000

It should be immediately obvious that the 1's show the subnet, and the 0's show the device number.

And so we come to a more general rule:

The Subnet Mask denotes the "Subnet" part of an IP address with a 1, and the "Device" part with a 0.

If we look at other subnet masks we can see this quite clearly:

255.255.0.0 = 11111111.11111111.00000000.00000000

We can also get into more exotic subnet masks:

255.248.0.0 = 11111111.11111000.00000000.00000000

It's very rare to find a subnet mask that isn't a combination of 255's and 0's for the simple reason that it is practically impossible to work out the subnet/device number of a "split" subnet mask.

Slash Fiction

Since writing 255.255.255.0 constantly is a bit of a pain, IT people have come up with a quicker way of notating the subnet mask of an IP address.

Normally, for our Finance Server in the example above, we'd have to write:

IP: 192.168.100.5

Subnet Mask: 255.255.255.0

But let's look at the subnet mask of 255.255.255.0 in binary again:

11111111.11111111.11111111.00000000

We can see that this is simply twenty-four 1's in a row. So, we can write the full IP address of the server as:

192.168.100.5/24

The "/" at the end of the address denotes the subnet mask; in this case, twenty-four 1's (or, in decimal, 255.255.255.0).

If we had the subnet of 255.255.0.0, the Full IP address could be written:

192.168.100.5/16

as 255.255.0.0 only has sixteen 1's in a row to denote the subnet address.

I hope that this has explained the concept of the subnet mask. As always, please let me know if you think that this could be a better explanation!

(PS: I do know that there is also the whole subject of VLANs that offer a better explanation for how to separate networks, but I'll leave that for another time!)

Why Square Waves Matter

A while ago I wrote a post about what Square waves are, how they are formed, and how they are really just made up of a bunch of increasingly higher-pitched sign waves.

I'm sure that was more than enough graphs and lines to scare anyone into submission. That's not such a bad thing.

But one thing that I didn't get into is why square waves are important to the modern sound engineer.

Distortion - nothing's perfect
Whether you want it or not, Distortion is at the heart of every piece of reproduced music. It doesn't matter if that reproduction is "live" (as in a microphone running through a mixing console at a rock concert) or if it was recorded 15 years ago and played back on a CD player.

There are a number of different reasons for this, some of which (like the reaction of various electronic components like capacitors) will be left for later; there's just simply too much to cover.

What I'd like to cover is the linking between Square Waves and distortion, specifically in two areas: Gain and Quantisation.

Gain Management
If you've ever used a mixing console (or pretty much anything that shows you the audio levels in the standard green-orange-red format) you've probably kept on turning up the volume until you've got some pretty red lights blinking all over the place. If you ask any experienced engineer they'll tell you that this means that they are "saturating" the console's amplifiers, leading to "some nice compression" and/or distortion.

But what's actually happening here?

At the heart of every bit of audio equipment is a humble little circuit; the Operational Amplifier (Op-Amp). This simple circuit has a never-ending number of variations and uses, but essentially all we need to know is that it takes an input signal and some power, and outputs the same signal but with more volts.

Your stock-standard op-amp drawings. 

Okay, once again I can feel that this picture means nothing to you.  Never fear; let me explain.
 The top picture is how most of us think of amplifiers: you put a signal (+ and -)  into a circuit and you get some kind of output (shown here as the line out the right hand side of the Amp).

In fact, it's perfectly acceptable to only show the +ve side of the input.
Essentially what this image shows is what we mentioned above; you put a small signal into the amp, and you get a bigger signal out. What actually goes on inside the triangle is inconsequential.

However, the bottom picture gives us a little more information. Here we still see our inputs and outputs (U+, U- and U0). There's no difference there. But we now also see two new variables; Ucc and Uee. These represent the "power" that we are supplying to the amplifier.

In normal operation, the input signal is small enough so that the output (U0) is less than the total power that you are feeding to the op-amp (Ucc and Uee).

But if your input is too big, you can get to a situation where U0 needs to be bigger than Ucc. Since the op-amp has run out of power it stops amplifying and simply outputs the maximum power; i.e. Ucc.

Two examples of clipping; we're interested in the lower example

If you look at the image above, you'll see what I mean. The dashed "threshold" lines indicate Ucc and Uee; in other words, the maximum and minimum possible voltages of the output, U0. For the moment, we're only going to look at the lower example.

You'll notice that as soon as the output reaches the maximum (Ucc) or minimum (Uee), the waveform flattens out, no matter what the input signal looks like.

You'll also notice that this looks exactly like a square wave.

If you have a look at the previous article on Wave Theory, you'll note that a Square Wave contains the original frequency as well as a number of  harmonic frequencies. If we were to look at the above waveform on a spectrum analyser you would initially see a single peak (the input frequency). As the input increases beyond the Ucc threshold, you would start to see the single peak get shorter and more harmonic peaks growing up, as if out of nowhere.

As the clipped wave starts to look more like a square wave, you get more harmonics

These harmonics are the "distortion" that we are hearing. 

Since no amplifier circuit is perfect, you'll always have a bit of this Harmonic Distortion cropping up along the way. It's one of the measurements of amplifier power; Total Harmonic Distortion (or THD), and is expressed as a percentage. The percentage refers to the amount of power that is "lost" in the harmonic peaks. In short, a lower number is better - the number is telling you how much "false" information you're hearing due to the amplifier circuit.

Essentially, you can "fake" an amplifier's power rating by increasing the THD tolerance. An amplifier than can deliver 100W at 0.1% THD might be able to deliver 300W at 5% THD. That's because you're increasing the output of the amplifier, and also increasing the harmonics that are creeping into your original signal.

Is that a bad thing?

Distortion isn't necessarily a bad thing. Rock 'n' Rollers have been using fuzz pedals and distortion generators for decades. As with all sound-related things, everyone has their own opinion. But you also need to make sure that you're getting the right gear for the job. A 300W 10% THD amplifier might be perfect for your car stereo, but you might want a 300W 0.1% THD amplifier for your Hi-Fi speakers in your living room.

Quantisation Noise

The other, very similar form of distortion that I'd like to discuss here is Quantisation Noise.

As we saw in my PCM post, digital audio effectively segments up an analogue signal into a bunch of "stepped" signals.

A 4-bit quantisation (grey) of a sine wave (red)

Hopefully, it won't take you much imagination to realise that the grey, "digital" signal will have a number of harmonic frequencies tied up within its square-wave-like structure.

If not, then scroll back up and have a look at the last image showing the build-up of harmonics in the Square Wave example again!

Of course, just as we explained in the PCM article, the above representation is a gross simplification of a digital signal; even a 16-bit signal has millions of steps, making the harsh corners a little more manageable.

That being said, a digital recording will always have a certain amount of distortion, or "Quantisation Noise," inherent in the system. In low bit-rate recordings or transmissions (e.g. bad digital radio), this comes through as a watery-garbling of the audio signal.

Here is an example of an 8-bit recording, and then the noise generated as this is reduced in bit-depth to:

4 Bits (like the above image)

2 Bits

But does it matter?

I've just had a look at a couple of Analogue-to-digital converters in 16 bit terms. These list the THD of the entire converter (including the op-amps and the rest of the circuitry) at about -100dB. That equates to a THD of about 0.0009%. 

Whilst there is a lot of complaints about the "sound" of digital recordings (and yes, I do admit that there are differences between the two), the chances are that the THD of the amplifier is many, many times greater than the harmonic distortion that is caused by the "digital-ness" of the recording.

Wave Theory - Part 1

Apologies for the delay here. I started writing a number of things into this post, however it became a long, boring, technical marathon. So I'm going to break up these types of articles and post some interstitial things as well. I originally wanted to look at Quantization Noise, but there were so many concepts involved that I decided to start with the "Basics".


Superposition

Science and nature detest corners. Curves, even constantly varying ones, are much easier to deal with. This is because curves add together nicely through a principle called "Superposition." Long story short, curves can be added together and taken apart without any difficulty. 

If you've ever looked at music on an oscilloscope, you'll see a random squiggle that never seems to stop changing. Whilst it can make for an interesting visual effect, it's not very helpful. However, due to Superposition, we can break that seemingly random signal down into the component inputs. This is usually called the Fourier Function Transform, but you might be more familiar with the term "Spectrum Analyser." (I strongly advise not looking too deeply into it unless you really like maths.)

If you run a signal though a Spectrum Analyser, you change the random squiggle into a series of columns. Each column represents a frequency range; the height of each column denotes the amount that each particular frequency range is contributing to the original signal.

Input signals (Left) and the resultant Spectrum Analysis (Right)
Thanks Wiki Commons

Let's have a look at some pictures; because they are easier to understand than words.
The top picture is your standard sine-wave. Since there is only one frequency, the Spectrum Analysis shows us that information; one tall column. All of the power in this signal is contained within that narrow frequency range.

The second set of images shows static; a totally random signal at low level. If we look at the Spectrum Analysis, you'll see that the power is spread randomly across all frequencies. We'll get into the origins of static one of these days. Just not today.

The last set of images shows these two signals superimposed onto each other. At each point, the "height" of the two signals is added together to make the bottom left image. It's a bit hard to see in the image, but you'll note that the curve is no longer smooth. However, this is nearly impossible to tell by looking at the input signal.
However, when we look at the Spectrum Analysis, we can clearly see that most of the energy is still in that main frequency range, but there is energy spread out across the other frequencies.

This effect is normally called "Noise".

Repeating Patterns
There is one other cool thing about superposition.

Basically, any repeating pattern can be built up with the right combination of sine waves. Take, for instance, a square wave:

Once again, Thanks Wiki!

If you have a look at the above image, you'll see three lines.
The Red line is a "true" square wave. The Green-dashed line shows a Fourier Approximation of the square wave using 5 component waves. The blue-dashed line uses 15 component waves. These waves are superimposed onto each other, like this:

The left-hand images shows our four component waves and their relative powers. Just by looking at the left-hand side, we can see that the "Fundamental" frequency, which is the same as the frequency of the square wave, has the most power.

The second column shows the superposition, but without adding the waves together. The third column shows the resultant, superimposed wave. As we go down the list, it starts looking more and more like a square wave.

The right-hand column, again, is what you'd see if you put the signal into a Spectrum analyser. There's a couple of points to note here:

•  For a square wave of a fundamental frequency F, the frequency of the component waves (f) is as follows - f = (2n+1)F, where n starts at 0 and goes to infinity
•  The power of each wave drops off significantly as n gets bigger (or, to put it another way, as the frequency of the component wave goes up, the power of that component goes down).

It turns out that once you get past n=16 or so, the power in the higher frequencies is so low that it no longer matters if you include them or not.



Next Time
It is very tempting to plough ahead here and talk about why I just wrecked your mind with superposition, but I won't.

Here's a hint though, it has to do with square waves and noise, and why digital and analogue aren't all that different.

As always, please feel free to post questions in the comments section!

Pulse-Code Modulation - Music to our ears

Okay, so we've just spent a lot of time looking at how data moves around a network. I hope that you were all with me for the ride. If not then please feel free to comment on any of the posts and I'll answer your post.

For the next couple of weeks I'd like to start looking at the data that you actually put into those packets.

One of the easiest places to start looking at digital signals is the humble Pulse-code Modulation, or PCM, method of encoding analogue information into a digital signal.

Analogue vs Digital
No, I'm not going to get into the "aesthetic" differences between analogue and digital, save to say that the only instrument that you can trust is your own ears. If it sounds better to you, then it sounds better to you.

I would like, however, to clarify something quickly. An "Analogue" signal is a proportionate signal with no real limitation. The local air pressure around a microphone or a signal can be measured by a device and turned into an electrical signal that is proportionate to the pressure. The higher the pressure, the higher the voltage. The voltage is an "Analogy" of the pressure.
This is what we refer to as an "Analogue" signal. Most simple electronic devices will process and run on analogue signals.
At some point, everything is an "Analogue" signal; the pressure changes that reach your ear drums or the light changes that reach your retinas are "Analogue".

A "Digital" signal, however, is somehow encoded so that it is no longer proportionate to the original signal. Through some kind of electronic process the Analogue signal is broken down into symbols of some kind that are readable only by devices that use that format. Before we can interact with them again then they need to be converted into an Analogue signal again. This process is called "Encoding" (Analogue to Digital) and "Decoding" (Digital to Analogue). Combine "enCOder" and "DECoder" and you get CODEC... but we'll get to those later.

Pulse-Code Modulation
Anyone who's ever Google'd "Digital Audio" will have seen a picture similar to this one:

Let's assume that we are looking at the Encode (Analogue to Digital) side of things (although the process is exactly the same in reverse).
The red line is our input signal; a standard sine-wave. This could be anything;  an audio signal, the number of people that like or dislike the current Prime Minister... it doesn't matter. We have a signal that is changing as time goes on.

The analogue signal is continuous and unbroken.

Pulse-Code Modulation sets a value (shown above as 0-15) for each equivalent amplitude. To convert the signal into a digital one, we record the value of the analogue signal at the start of each of the time divisions shown along the bottom access. This process is called "Sampling" - you are taking a sample of the Analogue signal at each of the time divisions.

You'll note that on the image above you can see a difference between the continuous Analogue (Red) signal and the Digital (Grey) one. It looks like a lot, right? In fact, the small differences in images like the above are one of the main arguments used by Analogue supporters. However, there is something missing from this picture...

Bit Rate
The picture above gives you a pretty good look at what you'd see in a phone-line; a 4-bit system. A bit is a single binary "symbol". One bit gives two states; two bits gives four, three bits give eight and four bits give sixteen states.
The number of "bits" that a digital signal contains is referred to as the Bit Depth. In basic terms, the higher the bit depth, the better the quality.

Even your most basic audio (CD-quality) has a bit depth of 16 bits; or about 65 thousand different states. The Human ear isn't really able to detect that kind of resolution; it would be like trying to look at the millimetre markings on a ruler that was 10 meters away.

Still, there are higher bit depths; standard digital audio (AES/EBU-3, which I will cover in a future article) runs at 20 bits (1,048,576 states) or at 24 bits (16,777,216 states). At this point, you're pretty much splitting hairs with a 2000-pound bomb...

But there is another factor that affects the quality of the sound; the rate at which the audio is sampled.
As a general rule, you want to take a sample at twice the frequency of the highest frequency you want to hear. The label on a new-born baby reads 20Hz-20,000Hz, although by the time you've used an MP3 player and gone to a concert or two you will be lucky if you can hear above about 17,000Hz.
Thus, the main sampling rate used in digital audio is 44.1kHz (CD-quality). "Professionals" will use 48kHz, or even go as high as 96kHz. Once again; at that level you are recording detail that humans just can't perceive. It's like taking a photo in ultraviolet; it might look brilliant, but there is no way for us to see the result.





I would like to continue this article, but in the interest of keeping things concise I will hold off for the time being. Next article I will look at a couple of the strange effects of PCM, and how we avoid them.
But for now, I must away. Until next time.

Layers of the Internet - Part 4

If you are new to this area, or are looking for the first three parts of this series, please use the following links:
Layer 1 - Link Layer
Layer 2- Internet Layer
Layer 3- Transport Layer

To recap quickly, the Link layer is the physical connection between devices (be it wireless, electrical or optical).
The Internet Layer is the layer that deals with the virtual separation of the entire Layer 1 network into smaller, virtual networks, and also allows traffic to be routed from one side of the planet to another.
The Transport Layer deals with two main things; where the data goes inside the devices that receive it, and also how the packets are passed around the networks (e.g. error correction protocols - re-sending a packet that never arrived at the destination.


Layer 4 - The Application Layer
For a second, let's think about what we are trying to achieve with a data network. We are trying to move information (be it computer data, an image, live audio... anything) from one point to another. If we look at what we've done in layers 1-3, we can see that we've done nothing but move the data from one end of the earth to the other.

Layers! Layer 4 is like the river and the clouds above; they rely on the lower layers of the earth, but they don't really care about them

.
But before any data can be delivered, we have to actually work out what data we are going to transport, why we around going to transport it and then what we are going to do with it at the other end. Data is useless unless we actually do something with it. For example, you could send a friend a photo, but if they can't view it then there was no point to transferring the data in the first place.

This is where the application layer comes in. It is the layer of the Internet that interfaces with the Users, but it is also the layer where we actually change the information that we are sending.

But Layer 4 isn't all about the Users, there's a lot of background processes involved. Let's have a quick look at a simple internet action; checking this blog:

• Step 1: You open your browser and type "http://arts-comms.blogspot.com". This is the start of the process; you enter a small amount of data; 30 characters. It's not much when you think about it.

• Step 2: Your browser tries to find the address of this page. If you were going to send a letter to a friend you couldn't just write their name on an envelope, post it, and hope for the best. The "Human-readable" Universal Resource Locator (URL) is similar to you friend's name; you can use it to describe them and look them up, but it isn't exactly an address. So, before your browser starts downloading this page, it looks out for a Domain Name Server (DNS) that will translate the URL into a TCP/IP Address.This address combines the Layer 2 and 3 addresses for the webpage. In this case the IP part of the address is 74.125.71.132, and the port is 80. This could be written 74.125.71.132:80.

• Step 3: Your browser now requests a session with the blogspot.com webserver. Now that your browser knows where to look, it will send a request to 74.125.71.132:80 to ask if it can start talking to the Server. The server will normally check that there are no bans on your device, or any other reasons that it wouldn't want to talk to you.

• Step 4: The Server opens a port and allows a session to begin. Should everything in Step 3 check out, the server will allocate a port number (this can be random, or fixed; it depends on the server) and then sends the information to your browser, letting it know that it's all good to go.

• Step 5: Your Browser requests the Webpage. Now that the Server and your browser are talking, the browser finally asks for this page, sending along with the request any additional information that might be required (e.g. your login details, or which exact page you're requesting).

• Step 6: The Server retrieves the webpage from its hard drive, and then transmits it as a series of packets. If you were to send a entire webpage as a single packet, it would be huge, and that would slow it down through the Internet. So the Server splits up the webpage into a number of small packets, adds a bit of information to let your browser know the order of the packets, and then transmits them through the network.

• Step 7: Your browser assembles the webpage, checking for errors as it goes. There is always a chance that some packets will get lost in transit. However, since the server has added information for your browser, it will know if anything is missing. If something doesn't make it then your browser will re-request the missing pieces.

• Step 8: Close the Session. Once everything has been checked, and you're happily reading away, the browser sends a message to the Server to finish off the session that was opened in Step 4. Once this is closed the connection disappears from the internet.

So, from your 31 keystrokes (including the "enter" key at the end) your browser and the Server have been communicating at Layer 4. You'll notice that I barely mentioned the Layer 1-3 protocols; and that is because  at Layer 4, just like all of the layers below it, the lower levels are transparent. It doesn't matter that I'm on a wireless connection and the server is on a optical fibre connection; they are connected up to Layer 3, and that's all that matters.

I'll be doing little write-ups on the various protocols as I continue to write this blog, so please stay tuned to the Glossary. Once I write an article on each protocol I will link it from the Glossary.

Layers of the Internet - Part 3

Before I start I would like to make a minor classification.
The "Layers" to which I am referring are the Internet Protocol layers, as specified in RFC 1122. I feel this model is more appropriate to the topics that I cover in this blog.


For those of you that wish to delve further into Networking as a subject, please refer to the Open Systems Interconnection (OSI) model. This model is the one that you would study in a networking degree, however it is slightly too specific for the purposes of this blog.




Layer 3 - The Transport Layer
In the previous two parts of this section of the blog, we looked at Layer 1 - The Link Layer and Layer 2 - The Internet Layer. In those layers, we say that each device was physically connected to all others, but with a bit of technology, we could divide up that huge network into smaller little sections.

I'll admit, it's difficult to visualise the Transport Layer. Here's some model trains.

As we have now connected on Layers 1 and 2, we can assume that our two devices are talking to each other. At this point it doesn't matter how that happens, or how many switches the packets have gone through in order to get from A to B. The tunnel through the Internet has been made, and for all intents and purposes, it isn't broken until the connection is shut off.

Layer 3 tells us two things; how a packet will move through a network, and where it will go when it gets there. In order to understand this we will examine two of the main Layer 3 Protocols, Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP). And yes, if you noticed that the "TCP" looks very similar to the "TCP" in "TCP/IP", you'd be correct. Since TDP is the most common Layer 3 Protocol, and IP is the most common Layer 2 protocol, people refer to this model as the TCP/IP suite of protocols. However, due to the complexity of the protocol, I will start off with the UDP.

User Datagram Protocol
Take a moment to think about how many different network-enabled programs you are running now. On my machine I have two browsers with a few tabs in each, Skype, two email clients, a couple of automatic software updaters.... the list goes on. Now think about a lonely packet headed my way. It has the MAC address (from Layer 1) so it knows where my computer is. It has my IP address (from Layer 2) so it knows how to find me on the network. But once that packet finds its way into my wireless adaptor, where does it go? Is it a part of a webpage, or a message from a friend via Skype, or an email?

Both the UDP and TCP use a number, called a "Port," to direct a packet to the program that it needs to arrive at. Your network adaptor reads the Port number and forwards the data to the appropriate program for processing.

A UDP packet contains a bit of information that identifies it as a UDP packet and a Port number. That's about it. The advantage of this is that you have a smaller packet, which means faster transmission times. We'll look into this a bit later.

Transmission Control Protocol
Whilst a UDP packet basically only has the Port number, a TCP packet contains a lot of additional information. TCP packets can contain information relating to tracking and delivery protocols. For example, a TDP packet could identify that it was the 15th packet in a sequence, and that the receiving device should send notification of receipt.

This information can be used to make sure that packets that have been lost due to network errors can be re-requested and re-transmitted. Packets that take too long can killed off. Devices can send packets to each other to check the speed and path of the transmission.

All of this extra information makes the packets bigger, and hence slightly slower. However, in any non-real-time application this extra transmission time doesn't make much difference. Since very few transmissions actually require real-time transmission, the more Robust TCP is more common than UDP.

The main place we see UDP packets in the Arts industry is in audio/video transmissions, like Dante or AVB.  UDP also requires less programming, and thus it is more common in smaller applications, so you may come across UDP systems in some control software.

"Layer 3 Switches"
Normally a switch will read the Layer 2 information (the IP Address) in order to find out where to send the packet. As we saw in the Layer 2 article, this is used to prevent every device receiving every packet on the network. If a switch only does this, then it is known as a Layer 2 switch.

However, some switches can be programmed to read the Layer 3 information. Switches can read the TCP information to adjust flow control (e.g. if the switch is over capacity, it can request the sending device to slow down the transmission rates). This can also allow for the ability for a switch to send more important packets before standard packets (e.g. Telephone or real-time audio is transmitted before a webpage).

The layers of the Internet - Part 2

The Internet Layer
In the last post I explained the basics of Layer 1 of the Internet Protocol; the Link Layer.

The Link Layer is where all devices are physically connected, either by wireless (e.g. Wireless LAN or 3G) or by a wired (e.g. Ethernet "Blue String" or ADSL) connection.

Whilst this is a great concept, if every device were to receive all of the data transmitted on the Internet, then we would be slowing down the process beyond belief.

Enter Layer 2 - The Internet Layer

This is Layer 2

On a network drawing, a network was always drawn using standardised symbols. Since everything that was on the "Internet" was connected at Layer 1, it no longer mattered how things were connected. You only had to show that there was some kind of connection.

And so the network symbol for the "Internet" became a cloud. It was some ethereal entity, floating out of the reach of Network Administrators across the globe.

Unfortunately marketing departments caught hold of this analogy, and thus Cloud Computing was born. You can see the "Cloud", and it brings you good things (like shade, and rain for your crops) but you have little to no power over it. It's there whether you like it or not. 

So how does it work?

Packet-Switching Networks

In this previous blog post I explained the anatomy of a standard Internet Protocol (IP) Packet. Packets are the currency for IP networks, and indeed the entire Internet. Without repeating myself too much, they contain two main parts; a "Payload" (the data that you want to move around the network) and a "Header" (which contains the addresses relevant to the Data).

In order to understand how the Internet works, we are going to have to introduce our first specific piece of network hardware: the Switch.

The image above is that of a "Switch," and it is a common thing to be found in data centres across the world. However you are reading this blog, somewhere along the line you are connected to a Switch. It might be a little 4-port switch that came with your ADSL plan, or you might be connected to a commercial-grade switch (like the one above) at work.

Switches are the building blocks of the Internet, and they elevate matters from Layer 1 (Link Layer) to Layer 2 (Internet Layer).

Every blue or pink cable in the above image connects to a device; a telephone, a computer, a printer etc. This is the Layer 1 connection. You can tell they are working by the blinking green lights. The Orange cables connect those switches to other switches, which the connect to other switches... until they reach whatever destination they need to get to. These connections are called "Uplink Ports", as they are headed up towards the "Cloud".

When a packet is sent to a switch, it "opens" it up and reads the "Header" (not the "Payload"). In the "Header" is all of the addressing information that the Switch needs to send the packet to where it needs to go. If the destination address is connected directly to the switch, then the packet will be sent directly to that device. If not, then the switch will send the packet to the "Uplink" port, at which point the next Switch will repeat the same process until the packet arrives at its destination.

By doing this, Switches make sure that you only receive the packets that you need to read your emails, browse your websites, control your motors, or route audio. Switches don't care what your packet has in it, so long as the address in the Header is valid.

The address used by the Internet Layer is the Internet Protocol (IP) Address. I will go into (much) more details about IP Addresses in a later post as the topic is as broad as the Internet itself. Suffice to say, a common IP address is an 8-byte address, usually rendered in four groups of numbers from 0 to 255, e.g. 192.168.0.254.

Once two devices are connected at Layer 2 they are considered "Networked" and can now communicate as if they were in the same room. Layers 3 and 4 deal with how they communicate, and we will cover these in the next blog post.

The Layers of the Internet - Part 1

The Internet is a complex place. There are many articles about the history of the internet, and it's a little beyond the scope of what this blog is about, so I won't go into it too much.
Suffice to say, the only way the internet works is through rigorous adherence to global standards. These standards were developed over the 15-year genesis of the Internet from research labs to commercial use.

One of the best ways to describe how these standards, and indeed the entire internet, works is known as the "layer" method.

Layers are a great way to explain many things, and the layers that we will be looking at today not only apply to networking, but to almost every form of computing or digital signal processing.

There are four layers in the "standard" Internet topology:

• Layer 1: Link Layer. This is the physical link between devices
• Layer 2: Internet Layer. This is the "virtual" layer where the data moves around networks
• Layer 3: Transport Layer. This layer defines how data moves around devices
• Layer 4: Application Layer. This is the layer that shows how data is shared between programs.

Those short little descriptions probably mean very little to most of you, so let me break it down a little bit more.

The Link Layer
When I think of the best way to describe the Link Layer, this image comes to mind:

What you're looking at there is a fairly typical "Distribution" switch, and a lot of optical fibre.
The Link Layer is the only physical connection layer in the Internet Protocol. It defines all the different ways that you can connect devices together if you want them to be on the internet.
If you put your mind to it, you could easily rattle off a lot of the different standards that are in the Link Layer, for example:

• Wireless Networking 
• Ethernet networking (a.k.a. "Blue String" - those blue cables that we are all familiar with)
• Fibre Networking 
• ADSL (Asymmetric Digital Subscriber Line - The way most of us get our Home internet)
• 3G/HSDPA - The wireless Broadband that most of us use on our phones.
• DOCSIS, a.k.a "Cable Internet" - networking over Coaxial cable, similar to Cable TV (Thanks djzort)

The list goes on. The greatest thing about the "Layer" system of the Internet is that it doesn't matter how you connect devices together at the Link Layer, so long as they follow the Link Layer Standards. As soon as you have that "blinking light" that shows you are connected then you can start passing information around the Internet Layer.

The Link Layer extends across the entire Internet. Just ponder on that for a second; every device that is connected to the Internet is in some way, shape or form, connected. The Link Layer is the only layer at which every device is connected; once you start moving into the "Virtual" layers (layers 2-4) you start segregating devices into separate virtual networks (or "subnets"). But for now, let's just muse on the topic of every device acting together in synchronism.

The last thing that I will mention about the Link Layer is the address that applies to it. Obviously there is no point in connecting every device on the planet unless you knew which one you wanted to talk to. Therefore every device that connects to the internet has a unique address. This address, known as the Media Access Control (MAC) address, is a unique number assigned by the manufacturer.

The MAC Address is a 48-bit number (that is, 48 "1's" or "0's"), which makes for about 300 Trillion different addresses. Every single device that is capable of connecting to a network has a MAC address. This laptop, for example, has two addresses; one for the Wireless connection and one for the hard-wired connection. Even so, the IEEE doesn't expect that we'll run out of MAC addresses this century.


We'll look at the Internet Layer next time (probably in a couple of days). I thought it best to break things up for now.

Time Division Multiplexing - TDM (the root of all good)

Rise of the Packets
In this post, I looked into the concept of Packets. I have a great fondness for packets, as it is packets that move the majority of the data in the world.

They are really good at moving data around in an ad-hoc fashion. Any piece of data can be broken down, transmitted and reassembled without any distortion.

Fall of the Packets
But that process is slow. It can take milliseconds to disassemble the data into manageable chunks, then add in all of the extra address, protocol information... and the data hasn't even left your device yet.

Obviously, when it comes to emails, images, or text a delay of even a couple of hundred milliseconds isn't going to be noticed by a mere human. So once again, the packet wins.

Time-critical Applications
However, let's take audio. I will go into how digital audio works at a later stage, but for now you should know that digital audio (e.g. CDs, common live audio mixers, MP3s etc) is made up of 24-bits of data, give or take.  (a "bit" is a 0 or a 1). This is repeated around about 48 thousand times a second; or once every 0.02 milliseconds.

So, if the a "packet" of audio data arrived milliseconds out of sequence, we would end up with noise. And not the good type; I'm thinking Lou Reid's Metal Machine Music.

A Saviour in time
To get around this we need a faster method of transport. Enter Time-Division Multiplexing!

Modern computing equipment is more than capable of processing things at speeds fast enough for "real-time" applications like audio. A 2.66 GHz processor (like one of the 8 I am using now) can process a "sample" (that 24-bit thing I was talking about before) in about 0.3 nanoseconds.

For those of you that don't like maths, that's about 65,000 audio "samples" per "Sample"; meaning that any one of my 8 processors could play music for me and still only be working to 1/65,000th of its capacity.

Okay, so those numbers may have caused you to glaze over. Hence the "bird on the wire" to liven things up.

The thing to take away from the maths above is that even a fairly stock-standard computer takes one look at time-critical audio and laughs it off.

So, how can this help us with faster data transport?


Enter Time-division Multiplexing
The simplest way is to give each "device," or data-source, an "address" that corresponds to a certain time.
This "address" will then offset the data from each device by a certain time; device 0 will have a 0s offset; device 1 will have an offset of 0.02 microseconds, device 2 will have an offset of 0.04 microseconds... until we get to device 65,000 something.

Let's have a look at the diagram below:

Here we only have two devices; Red and Green. Green has address "0" and Red has address "1".
First up, the Green wants to transmit some data, so it goes into the first "timeslot" (a timeslot is the base unit of a TDM network; kind of like a packet). Since there is no "Red" data in the first timeslot, the next slot is blank.

However in the second "Timeslot" both Green and Red want to transmit something. Green goes first and transmits its data, and then Red transmits, so we see a Green block and then a Red block.

And so on; each device waits for its turn, then transmits the data.

To read the data you need to know which data you are looking for, wait until that "timeslot" appears, and then read the data.

Forgiving the Chinese characters, this image shows the same concept, but for more devices. On the left we see all of the data that we want to transmit. In the middle we can see that each device waits until it is their turn, then transmits like crazy.
On the right hand side we can pick out any particular timeslot and read the data.



Why should we care?
One thing that you may have noticed in the above is that all of the devices have the ability to read all of the data on the network.
Time-Division Multiplexing (TDM) Networks are relatively simple; each device is directly connected to the next, and will usually transmit the entirety of the data to all connected devices.

One application that really likes this is large matrices. For example, take a communications. You could have 100 people trying to talk to each other. Any one of those 100 people may want to talk to any (or all) of the other 99 users at once.
A TDM Network makes mincemeat of this; each user is allocated a timeslot on the network. When they talk, their audio fills up their timeslot. When they aren't talking, nothing is transmitted (remember our Red and Green above).
At the other end, the user receiving the call pulls out the audio from all of the timeslots that they want to listen to.

Of course, this requires a bit of computer processing, but most common communications networks are more than capable of handling thousands of users on one TDM network.

In fact, most matrix routers/switchers employ a TDM network of one kind of another. The concept of a TDM can be applied inside a single "box". For example, a video matrix. Each input takes up one timeslot; each output reads from one timeslot. In this way, any input can go to any output without causing interruption to the video. When a TDM is contained in a single box, it is usually referred to as a backplane.

Downfalls?
The biggest drawback of a TDM  TDM network is limited not by the address space, but by the processing speed of the devices and the size of the data being transmitted. Transmitting audio through a computer's processor gives many thousands of audio channels, however transmitting high-definition video over standard networking architecture yields only tens of channels.

TDM networks need to be connected to all other devices in the network. There are some funky devices out there that will bridge two TDM networks, but these are usually limited.

TDM networks are also super sensitive to clocking. Once again, I'll go into clocking in a different article, but for the time being think of it this way.
Think of all of the timeslots as trains. However on TDM station there aren't any "The next train goes to Sydney" announcements, there is only a timetable.
If you only have your wristwatch to go on, you may catch the wrong train; how do you know that your 1:13 is the same as the railway's 1:13?
However, if there is a clock on the station, and one on the train, then you can confirm that your train is the right one before you board.

The same thing happens with TDM; unless you synchronise all of the devices you have no way of knowing which "train" (timeslot) is the right one.


Conclusion
I hope you have enjoyed this. Please comment if you don't "get it" and I will edit this post in order to make it easier to understand.

TDM networks are prevalent in audio, video and communications networks, and that's why I wanted to get this up earlier rather than later. Apologies for the maths earlier; I hope the bird picture helped you through that.

What is a Packet?

Packet-switching networks, or simply Packet networks, are the most common form of network around.

But what exactly is a "packet"?

Basically, a packet is a bit of information, formatted to move around its specific network. It usually consists of two parts; a payload and formatting/addressing information. The payload can be anything that can be digitised.

For those of us coming from the Audio side of things, we're already used to the thought of packeting information. Think about a Digital audio stream. We find out the amplitude of the analogue audio signal and convert it to 1's and 0's. But before we send it around our mixer, CD player or whatever, we also put a few extra bits of information into it; how many samples per second, if it is a stereo or a mono channel, etc.

The are common examples in Lighting as well. A DMX signal, when looked at obliquely, is a packet. It has a payload (a value) and addressing information (a DMX channel).

However, the most common packet, the one that causes the most joy and woe in the world, is the Internet Protocol (IP) packet.

Below is a pictorial description of the IP packet. Instead of trying to bore you with details, I thought this would be an easy way to show what I mean.

By Nicolargo (Own work) [GFDL (www.gnu.org/copyleft/fdl.html) or CC-BY-SA-3.0-2.5-2.0-1.0 (www.creativecommons.org/licenses/by-sa/3.0)], via Wikimedia Commons

I can hear your brain cogs turning trying to understand what it is that you are looking at.

Let's start at the bottom. The black section that trails off toward the bottom of the page is the data that we are trying to transmit. This can be anything digital; a photo, some audio, a DMX channel change, a webpage... so long as you can turn something into 1's and 0's, you can transmit it over an IP network.

I think I should say that again; anything that can be digitised can be sent over an IP network.

That is why these networks are so powerful; any data, no matter what it is, can be sent over these networks. The amount of data contained in a packet has no real limitation. If you have a large bit of data, you can break it down into a number of packets, send it through the network, and reassemble it at the other end.

As we move from the bottom to the top, we see some things that should appear obvious; Options, Source and Destination address. Normally, a device that receives a packet will read this information and decide what it has to do with it.

I will be going into the way the internet works in future articles, but for the time being please think of a mail sorter in a post office. He looks at each letter and then sends it off to the next closest post office.
That's how addressing works; network equipment will look at the addresses and then decide what to do with it.

Next up the line is a bunch of information that doesn't really matter to us... yet. There are basic housekeeping flags, like the total length of the packet, the time that the packet should be kept around (time to live) and some checksums.


The reason that a lot of Performance Networking protocols (like Cobranet) don't work well on combined networks is because they don't have all of this "extra" information. They have the Source and Destination addresses, but they don't bother with the rest of the data.

In the "olden days" of networking , you could get away with this. Networks were kept separate, and in many cases all of the data was transmitted to every machine on the network.

However, as time has gone on and the Networking world has taken leaps and bounds ahead of the performing arts (private money helps...) they have added a lot of smarts to their technology. That is why we need those extra checksums, version and protocol type flags.

As mentioned above, the next few posts will be looking at the way packets move through the Internet, but please, if you have any questions, just email me and ask.

N.B. As mentioned in the comments below, a "Packet" is technically only the information that is shipped around at Layer 3, the "Transport Layer". This is why technologies such as CobraNet are not strictly IP-based, and also why they do not work on enterprise-type networks. However, this is a little beyond the scope of this Blog, which hopes to examine Networking in Theatres/Broadcasting, as well as the Theatre/Broadcasting aspects that are relevant to a Networking technician working in those areas. Please forgive any shortcuts that I may be taking in order to clarify a point.