Voice and video have used digital networking for a while now,
so there's a lot of good detailed information out there.
However, a lot of these guides assume you already know the basics.
So in this video, we're going to build that basic understanding
of voice networking from the ground up.
The following two videos will take that and create a really solid foundation.
We're going to start by taking a look at where digital
phone systems have come from
in the really old days, phone calls went through a switchboard.
This required an operator to connect you through to your destination.
The building they were in was called a telephone exchange.
Smaller regional exchanges are called branch exchanges.
If a company was large enough, they would have their own
switchboard and operators for calls within the company,
as well as for connecting external calls.
These were called a private branch exchange, or PBX.
Exchanges still exist today,
but they're very different to what we're seeing here.
Of course, this was ultimately too expensive.
So this became automated.
We're looking at an example of a larger exchange here.
But this was also true for private exchanges.
This led to the private automatic branch exchange
or PABX. Developments like this meant that every desk could now have a phone,
but there was no need to have a separate phone line for each desk.
The basic function of the PABX is
the same, it routes a call from one location to another.
But over time, the PABX got fancier and offered features
like call forwarding, voicemail, voice menus and things like this.
These days, the terms PBX and PABX
are used interchangeably.
These phone systems connected to external exchanges
using copper phone lines, just like you may still have at home,
but they had a lot more of them, of course.
The copper phone lines
connected the business to a local telecom providers exchange.
Calls into and out of the premises would route through this exchange
then probably through more exchanges until the call
ultimately reached its destination.
This network of copper phone lines between exchanges, businesses and private
residences is called the Public Switched Telephone Network,
or PSTN.
So far, this is all been using analog electrical signals,
but then digital links like ISDN started to come along.
Each ISDN link was a dedicated 64 K circuit between the provider
and the business.
Each of these circuits could carry a single phone call.
This is effectively where we started mixing phone calls and data
on the same medium because ISDN could also carry data.
In fact, we could bundle several ISDN links
together to carry more phone calls or create a faster internet connection.
PBX's were getting smaller and more manageable, and now we're able
to integrate the older analog lines with the newer digital links.
Sometimes it meant translating between the two types.
Phone networks came first.
Each desk had a phone on them when computers
were still only for a few users when data networks caught up.
They were entirely independent of the phone network. This
usually meant separate administrators, separate cabling and separate hardware.
This is clearly not as efficient as it could be.
So eventually we started to see unified communications or converged networks.
The whole point of this is to combine voice and data onto a single network.
And that's where we are today.
We have phones connecting to switches, digital phone
and data lines and IP enabled PBX.
In general, we call this voice over IP or VoIP.
Something we're seeing more and more is the addition of video
to this, as well as not just voice anymore.
At the heart of this voice network is the IP PBX.
But practically, we just call it the PBX.
This is like the traditional PBX in that it routes calls from one location
to another, both within the building and externally.
The difference is the PBX now has an IP address as it's in the data network.
It may be a physical or virtual appliance in your server room,
or it may be hosted by a provider in the cloud.
Some physical appliances support special hardware to connect different
types of phone lines like SIP trunks, which we'll talk about soon .
ISDN and PSTN.
But there are other advantages to putting the PBX onto the data network.
For example, they can now integrate
with Active Directory to build a phone number list.
We can add new applications to add more features, such as dialing
a number from your PC, and we can send voice mails to an email address.
It's also common
these days to have a mixture of hard phones and soft phones.
If you're not familiar with these terms, a hard phone is a physical phone
that sits on your desk.
It connects to a switch with a regular data cable.
A softphone is a software
based phone installed on your PC.
You can use a headset with this or a microphone and speakers.
The soft phone uses the PC's connection to the data network.
Regardless of the phone type, each phone has an extension,
extensions are usually a number, but in some cases
it could be something else, like an email address.
The PBX maintains a list of extensions and how to reach them.
This is how it can route a phone call from one phone or PC to another.
These extension numbers are often
linked to a phone that can be dialed from outside the organization.
These are called DID or DDI numbers.
As far as I know, they're exactly the same thing. One term
is more common in Europe, while the other is more common in America.
These phone numbers need to be unique internationally,
so these phone numbers are assigned to your organization
by your voice provider. To get these external calls working
The PBX has one or more phone lines to an external provider.
ISDN is an example of this, but it's getting less and less common by the day.
These days, SIP trunks are most commonly used.
SIP trunks, a different to ISDN
as they're regular routable data links like a WAN or internet connection.
They use IP based protocols
to carry voice traffic between your site and the provider.
one of these protocols is called SIP, which is why we call these SIP trunks.
Previously, the number of phone calls you could have
once was based around the number of ISDN or PSTN links that you had.
Now it's based around two completely different things.
These are the bandwidth of the SIP Trunk
and an agreement you have with your provider.
For example, you may be paying them to allow five
calls, 500 calls, 5000 calls, whatever it might be.
The whole point of this section is that an organization
can have many phones, each with an extension,
extensions often have a DID linked to them, although they don't have to.
And we no longer need separate phone lines for each call.
Let's think a
little about how this might work in a small business.
Here we have a small network topology that you might see in a small organization.
We'll take a look at a much bigger one in the next video.
first, you can see that the desk phones are connected to regular network
switches. The port they connect to, is configured with a voice VLAN
that is a VLAN used solely for voice traffic.
These phones do not plug into the wall for power.
Instead, they draw power directly from the switch,
using power over ethernet or PoE.
PoE enables the switch to deliver a small amount of power
just enough to run the phone, in fact. Not all switches have PoE
So keep this in mind if you're buying new switches.
The phone can also get basic information about the network
as well as advertise its own features directly with the switch.
It does this by using either CDP or LLDP.
CDP is a Cisco only protocol, while
LLDP is supported by, well, pretty much every vendor.
Now, if you're planning a voice network, you might think
that adding phones will double the number of switch ports you will need,
after all, you will need one for the PC and one for the phone, right?
Fortunately, no, it doesn't work that way.
Desk phones have a small three port switch inside them.
one port connects to the upstream switch.
The second port is internal and connects to the phone hardware,
and the third port is for the PC to connect to.
This means that you still only need one switch port per desk.
Does this mean that we are now
mixing our regular data with the voice traffic?
Also, no.
The upstream switchport is configured with two VLANs.
one is the voice VLAN, which we spoke about earlier
and the other is the regular data VLAN that the PC connects to.
So the phone is connected to the network and it has power.
They get some basic information from the switch using CDP or LLDP,
but that is not enough.
It also needs an IP address and needs to learn about the PBX.
So for this, it uses DHCP.
The DHCP
server assigns an IP address to the phone just like it would do any other device.
However, the DHCP server is also configured with some additional options.
In addition to the IP address, the DHCP server
gives the phone a TFTP server address, which is often
the PBX itself, where it can download a configuration file.
This file
contains all the other information the phone needs.
As mentioned earlier, the PBX is the heart of the phone system.
It can integrate with Active Directory to get a list of users
and their phone numbers.
When a phone comes online,
it will register itself with the PBX using the SIP protocol.
In this case, the phone is the SIP client and the PBX is the SIP server.
We look at SIP in detail in a future video.
Everything's ready to make a phone call now.
When one phone wants to call another, the calling phone
will send a signal to the PBX using SIP.
The PBX knows the IP addresses of all online phones on the network
because they all registered with the PBX when they started up.
In addition, the PBX has a list of dial plans.
These are a list of phone numbers and extensions
and information about how to route calls there.
Dial plans are configured by an administrator
and can be simple or quite complex, depending on what's needed.
Using this information, the PBX can then signal the destination phone.
Also using SIP, it gives each phone information
about the other and tells the destination phone to ring.
If someone were to pick up the phone at the receiving end, voice traffic
will start to flow directly between the two phones using the RTP protocol.
Notice of the PBX is not involved here.
It is only used for calls signaling and setup.
You can probably see how the PBX is still a bit like the phone operators
of the old days
when the phone call ends, the phone signals the PBX once
again to keep it up to date.
What happens
if we want to place or receive external phone calls?
This is where the SIP trunk comes into play.
Remember that this is just a regular data link that connects the business
to their phone provider.
The SIP trunk logically connects our PBX to our providers PBX.
When a call needs to come into our organization,
the provider will send signaling information to our PBX using SIP.
The PBX brokers this connection to the destination phone.
When the call is accepted, the voice data
using RTP travels between the local phone and the provider.
However, there are cases where the PBX or another
device will proxy the RTP traffic on the phone's behalf.
We'll see some cases of this in the next video.
Now, since we're talking about voice data, we should think
about what voice really is.
It's a
series of sounds which by nature are analog.
However, we're talking about digital networks, so how do these work together?
The answer is through codecs.
a codec is a software module that converts
sounds into digital information and back again.
There are many different codecs out there, and they all have complicated
names like G 711 and G729.
They run on the phones themselves
and sometimes even fax machines.
You're probably wondering why there are so many different options to choose from.
The reason is that they focus on different things.
Some will focus on high quality audio, while others
will focus on optimal bandwidth use.
The important takeaway here is that the two phones need
to support the same codec if they are to work together.
There are devices that can translate
between two codecs, which we'll discuss in the next video.
Also, keep in mind that your voice provider will only support certain codecs,
so when you buy a sip trunk from them, they'll give you a list
of what they support and which ones they prefer you to use.
So that's the start of our introduction to voice series.
There are two more videos to go in the next video.
We're going to look at more advanced components
it would find in a larger or enterprise network.
I hope to see you there.