Intro to VoIP Telephony (Part 1)

Welcome to VoIP Telephony training, as part of our QSC Quantum Training,

an advanced service and troubleshooting curriculum.

My name is Patrick Heyn and I’ll be giving you this brief overview on VoIP.

I'm not gonna lie, this is gonna be dense.

So buckle up, don’t be a hero, take breaks when you need them, and let's get started!

At its most basic, Voice Over Internet Protocol (VoIP) is a group of protocols.

We are going to focus on the one that most people use today, which is SIP.

VoIP is a method of sending audio over an IP network instead of a circuit based network like PSTN.

There are a bunch of different protocols that work with SIP like SDP, DTMF, RTP, and audio codecs.

We're going to dive into all of these as well.

The first one is the PSTN network. That sits outside your enterprise and makes up the telephony backbone.

That network is all digital now as well.

SIP stands for Session Initiation Protocol. It's a signaling protocol so it doesn't actually send audio.

It's a method we use to control, set up the call, and register with the call server.

We also have Session Description Protocol (or SPD).

This is used in a variety of technologies and it's not just VoIP, like video applications,

session announcements, invitations, and parameter negotiations.

The Real Time Transport Protocol (or RTP) is used to transport all multimedia

(including audio, video and other data)

over a packet switch network, which could be over the internet or over a private network.

SDP also negotiates the audio codec,

which are the algorithms used to encode and digitize the audio, each of which have its own benefits.

Dual-Tone Multi Frequency (or DTMF) is the protocol for how we send digits over the network.

The PSTN network is mostly digital,

and the only remaining analog portions are the local loop which are the things like your landlines.

Make no mistake, landlines are going away but many people still do have one.

Here we have a basic diagram of the telephone network and various components.

On the left you have your corporate network which might include a PBX or similar device like a call manager.

In the middle you have the PSTN and then on the right you also have celluar networks.

VoIP consists of two main components.

The signaling plane and the bearer plane. Let's keep those in mind for later.

We use SIP because…. well, because it's most popular, and it’s a well defined protocol.

There are many RFCs that describe SIP.

Our SIP engineers at QSC that write the software for our softphone follow the RFCs.

They are also available for anyone to look up and check if someone is following SIP protocols.

The problem is that people are allowed to interpret them however they deem fit.

The primary one is RFC 3261, but as you can see, there are many others.

SIP can use TCP, UDP or TLS. UDP and TCP are not secure connections and TLS encrypts the SIP signaling.

The proxy or call server is the device that we need to register with, also set up or receive the calls.

It also keeps a record of all the endpoints that are registered with it

and routes the call in and out of the proxy.

Each individual device registration is known as the line,

extension or directory number, and depending on the SIP service you use, carries it’s own ID number.

The server or proxy will either be an on premise server or a hosted service provider.

On- premise devices like Cisco Call Manager and Avaya are still very common

but hosted providers like Ring Central are quite popular nowadays.

If you use a hosted provider then you'll need to be able to reach the internet from the LAN you are using.

Before you get started on a SIP registration, you’ll need the proxy address

(or a fully qualified domain name of the server),

the line ID (sometimes called the extension or the directory number), and the password.

Now, some devices like Cisco Call Manager require a few other things.

You might need to have a username or digest credentials, and in some cases,

the username or authentication ID may be different than the extension.

Within SIP, there are messages called requests and responses.

You don't need to memorize all of these, but let’s take a look at a couple of them.

The INVITE is the message to start the call, and the BYE message ends the call.

If you need to end a call before a successful connection, the you get a CANCEL message.

There are only 14 types of requests but far more types of responses.

Responses can be categorized into various groups.

There are provisional, successful, redirection, and a few types of failure responses.

Here are some of the more common responses.

A “100 Trying” acknowledges a call phone request and indicates that the server is processing the request.

The client will send the invite and the server will respond with a trying.

A “200 OK” indicates a successful registration when you make a call.

A 401 unauthorized seems like it would be an error but that is part of the SIP protocol.

500 series messages represent a failure of some kind. Normally those failures would be on the server side.

For example, when you see several 503 errors in Cisco Call Manager,

this usually means that something is not set up properly.

Here’s a phone registration example from the RFC.

First a registration packet is sent and the server responds with a 401 Unauthorized.

Now, that is normal and by design.

It’s just trying to register without a password and that's how we're supposed to do it.

That's how the SIP protocol works.

The server comes back and say there's no password and sends us a hash which we will use to hash our password,

and then sends it back. The second REGISTER F3 message sends the password.

And then the “200 OK” verifies a correct password and registration.

At this point, if we see anything else, like a 503 or other error,

then we are going to show that fault in a softphone.

If you do a network capture this is what it's going to looks like in Wireshark.

The request is a registration request and the destination we are sending it to, asking for a response.

Our transport type in this case is UDP.

Once we get the 200 OK the green light goes on in the softphone, and we are ready to make calls.

You can show this in the call flow form in Wireshark and it gives you a bit of a better view of what it looks like.

At this point we are registered with a server.

When we get back from a break, we’re going to take a look and see how to make those calls in and out.

We’ll see you when you get back!