What is VoIP

VoIP (Voice over Internet Protocol) is a very small name that covers a wide range of techniques for making phone calls over an IP based network.

The largest IP network is the Internet, VoIP makes it possible to have a life conversation with someone on the Internet, in the same country or at the otherside of the world. So it is an addition to the e-mail and chat (MSN, ICQ, etc.) functionalities.

How it was

It all started with Bell and with the POTS (Plain Old Telephone System). With POTS I mean a telephone connected to a wire that runs to the wall and then the magic starts. When you dial or touch a number you get someone on the other end and you can talk to eachother. That is POTS for you in a nutshell.

In this section we are going to explain a bit of the magic. This will help you better understand everything involved in VoIP networking. The following picture shows a more detailed picture of how a telephony system works.

The first new element is the PBX which is short for Private Branch eXchange. A PBX gives you the option to connect more phones to the line you get from your local telecom supplier, ofcourse only one single phone can use the outside line, but you now have the choice of which phone you use. It also gives you the ability to call from one phone to the other without using the telecom suppliers line. So it is free of charge.

The PBX can also be used to have more lines coming from your telco (Telecom Company). But more lines means more costs. A solution to this problem was found in TDM (Time Devision Multiplexing). TDM sends more phone calls across a single line by using so called time slots. Every conversation gets a time slice on the link. When that happens fast enough your ears will not hear that the conversation was cut into pieces.

The TDM signal is then send across the PSTN (Public Switched Telephony Network) using a trunk line. Actually the trunk line is the connection between two telephone switches. And the line between your closed telephone switch and your telephone is called the Local Loop. The fact that it is called a PSTN, already indicates the fact that we are talking about switching, a PBX is in it's most basic functionality nothing more than a switch. A switch that switches between different lines and telephones.

Why change?

Price. Since most people in a western world do have a broadband connection to the Internet and the amount of time it takes to send a packet across the Internet (latency) to another computer is so little, voice applications are possible. Since everybody is already paying for her or his DSL-link, why should we also pay for telephony, while we can use the same link for that?

Another reason might be flexibility. The old telephone system tied a number to a phone. If you where not near the phone, you would miss the call. Mobile phones have solved this problem. But it didn't solve it for those areas where your mobile phone provider does not have an antenna. VoIP might be a welcome addition in those areas. If there is Internet, but no mobile phone system, like in some developing countries, VoIP might just be the solution.

A VoIP call

As with normal phone conversations VoIP also consists of two functions, which you normally take for granted: First you have to be connected, before you can talk.

VoIP is very different from a normal phone call. Where the telecom operators are already used to digitize voice before it is send across the large cables they operate, all is send in a nice and orderly fashion. The problem with IP is that it promises to make an best effort to get the data across. It doesn't promise it gets it there, nor does it promise to get it there intact, and even more troublesome for our voice call it doesn't promise to get it there in the order in which the data was send.

All these parts will be discussed in the document. We will start with the analog to digital conversion, then discuss how we can get the signalling right and last we will discuss what techniques can be used to make sure phone calls can be made over such an unreliable network.

PCM: Pulse Code Modulation techniques

To convert the 4 kHz voice signal of a phone call t a digital format we need a sampling rate of 8 kHz (for Nyquist) or 8000 samples per second. Encoding that into a 8 bits representation, means that 8x8000=64.000 bits per second need to be used to achieve near perfect voice encoding. This is what is called PCM encoding, which is the most used standard in the telecom world, the 64 kbps time slot.

μ-law for North America and a-law for Europe are variants of PCM used to code the analog signal on a logarithmic scale using 12 or 13 bits instead of 8 bits (see Standard ITU-T G.711)

Out of this basic standard all kinds of different techniques grew to create standards that used less bandwidth, but maintained more or less the same quality.

CELP: Codebook Excited Linear Prediction techniques

CELP uses a very complex scheme of encoding through which it is able to produce voice streams of less then 16 kbps. The downside is that delay is added through the extra processing needed. G.728 is an good examle of a CELP codec, which has an delay of less then 2 ms.

Which codec?

The codec that best suites your needs is highly dependend on the type of application you are using. With every decision you win on one point and loose on another. The only thing I can do is to give some general considerations.

First we have to consider what the most important factor is: quality or bandwidth.

Read more

http://www-mobile.ecs.soton.ac.uk/speech_codecs/

http://www.speex.org/docs.html

http://www.ilbcfreeware.org/documentation.html

http://www.shoshin.uwaterloo.ca/~jscouria/GSM/index.html

Protocol suites

At the time of writting there are two major protocols and one new commer that are describing the protocols used to setup a connection and how the voice data is transported to its final destination.

Within the protocol suites different approaches are choosen for call setup and tear down (signaling) and the transport of the actual voice data (data steaming). Luckily there are also all kinds of naming conventions, to make things more complex. To help you a bit we created an overview that hopefully clears the way in better understanding the different parts:

Signaling protocols are also refered to as session protocols, since they are responsible for setting up and tearing down a connection. We will decsribe them from old to new with both their pros and cons. Remember however that standards change with new versions, so always check if what is written here is still correct.

Gateway protocols (trunk)

Which protocol

Effectively the IAX(2) protocol uses the least overhead, the downside is that is not a open standard. It is created by the makers of the Asterisk PBX, so there might be little support for this protocol in devices.

H.323 is already considered absolete sinds the introduction of SIP, however there are still a lot of devices and applications that use the H.323 protocol so it all depends on the devices you have and that you want to support.

Codec delays

Codec table

Latency

It takes time to transport data across a network. The longer the distance the more time. Ethernet, Internet, VSAT.

IPv4 TOS

According to [RFC791], the IPv4 TOS octet is divided into a 3 bit Precedence field and a 3 bit TOS field. The last two bits of the TOS octet are reserved for future use:

The Precedence field is defined according to RFC 791 as:

QoS