Video Transcript
Intro to VoIP Telephony (Part 1)
7m 19s
00:07
Welcome to VoIP Telephony training, as part of our QSC Quantum Training,
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an advanced service and troubleshooting curriculum.
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My name is Patrick Heyn and I’ll be giving you this brief overview on VoIP.
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I'm not gonna lie, this is gonna be dense.
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So buckle up, don’t be a hero, take breaks when you need them, and let's get started!
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At its most basic, Voice Over Internet Protocol (VoIP) is a group of protocols.
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We are going to focus on the one that most people use today, which is SIP.
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VoIP is a method of sending audio over an IP network instead of a circuit based network like PSTN.
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There are a bunch of different protocols that work with SIP like SDP, DTMF, RTP, and audio codecs.
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We're going to dive into all of these as well.
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The first one is the PSTN network. That sits outside your enterprise and makes up the telephony backbone.
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That network is all digital now as well.
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SIP stands for Session Initiation Protocol. It's a signaling protocol so it doesn't actually send audio.
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It's a method we use to control, set up the call, and register with the call server.
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We also have Session Description Protocol (or SPD).
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This is used in a variety of technologies and it's not just VoIP, like video applications,
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session announcements, invitations, and parameter negotiations.
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The Real Time Transport Protocol (or RTP) is used to transport all multimedia
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(including audio, video and other data)
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over a packet switch network, which could be over the internet or over a private network.
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SDP also negotiates the audio codec,
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which are the algorithms used to encode and digitize the audio, each of which have its own benefits.
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Dual-Tone Multi Frequency (or DTMF) is the protocol for how we send digits over the network.
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The PSTN network is mostly digital,
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and the only remaining analog portions are the local loop which are the things like your landlines.
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Make no mistake, landlines are going away but many people still do have one.
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Here we have a basic diagram of the telephone network and various components.
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On the left you have your corporate network which might include a PBX or similar device like a call manager.
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In the middle you have the PSTN and then on the right you also have celluar networks.
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VoIP consists of two main components.
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The signaling plane and the bearer plane. Let's keep those in mind for later.
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We use SIP because…. well, because it's most popular, and it’s a well defined protocol.
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There are many RFCs that describe SIP.
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Our SIP engineers at QSC that write the software for our softphone follow the RFCs.
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They are also available for anyone to look up and check if someone is following SIP protocols.
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The problem is that people are allowed to interpret them however they deem fit.
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The primary one is RFC 3261, but as you can see, there are many others.
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SIP can use TCP, UDP or TLS. UDP and TCP are not secure connections and TLS encrypts the SIP signaling.
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The proxy or call server is the device that we need to register with, also set up or receive the calls.
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It also keeps a record of all the endpoints that are registered with it
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and routes the call in and out of the proxy.
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Each individual device registration is known as the line,
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extension or directory number, and depending on the SIP service you use, carries it’s own ID number.
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The server or proxy will either be an on premise server or a hosted service provider.
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On- premise devices like Cisco Call Manager and Avaya are still very common
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but hosted providers like Ring Central are quite popular nowadays.
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If you use a hosted provider then you'll need to be able to reach the internet from the LAN you are using.
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Before you get started on a SIP registration, you’ll need the proxy address
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(or a fully qualified domain name of the server),
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the line ID (sometimes called the extension or the directory number), and the password.
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Now, some devices like Cisco Call Manager require a few other things.
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You might need to have a username or digest credentials, and in some cases,
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the username or authentication ID may be different than the extension.
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Within SIP, there are messages called requests and responses.
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You don't need to memorize all of these, but let’s take a look at a couple of them.
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The INVITE is the message to start the call, and the BYE message ends the call.
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If you need to end a call before a successful connection, the you get a CANCEL message.
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There are only 14 types of requests but far more types of responses.
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Responses can be categorized into various groups.
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There are provisional, successful, redirection, and a few types of failure responses.
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Here are some of the more common responses.
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A “100 Trying” acknowledges a call phone request and indicates that the server is processing the request.
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The client will send the invite and the server will respond with a trying.
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A “200 OK” indicates a successful registration when you make a call.
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A 401 unauthorized seems like it would be an error but that is part of the SIP protocol.
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500 series messages represent a failure of some kind. Normally those failures would be on the server side.
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For example, when you see several 503 errors in Cisco Call Manager,
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this usually means that something is not set up properly.
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Here’s a phone registration example from the RFC.
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First a registration packet is sent and the server responds with a 401 Unauthorized.
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Now, that is normal and by design.
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It’s just trying to register without a password and that's how we're supposed to do it.
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That's how the SIP protocol works.
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The server comes back and say there's no password and sends us a hash which we will use to hash our password,
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and then sends it back. The second REGISTER F3 message sends the password.
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And then the “200 OK” verifies a correct password and registration.
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At this point, if we see anything else, like a 503 or other error,
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then we are going to show that fault in a softphone.
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If you do a network capture this is what it's going to looks like in Wireshark.
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The request is a registration request and the destination we are sending it to, asking for a response.
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Our transport type in this case is UDP.
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Once we get the 200 OK the green light goes on in the softphone, and we are ready to make calls.
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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.
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At this point we are registered with a server.
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When we get back from a break, we’re going to take a look and see how to make those calls in and out.
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We’ll see you when you get back!
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Intro to VoIP Telephony (Part 1)
7m 19s