VoIP (Voice over Internet Protocol), generally pronounced as “voyp”, is indeed the future of telephony. It’s more precisely known as internet based telephone system or broadband phone service. The term “VoIP” doesn’t really mean the involvement of a single element, but it consists of several elements that together produce a voice call over IP. You might already be using VoIP and not even know or realize it. Even I didn’t realize it in the first place when I used VoIP for the first time. Nowadays, VoIP is gaining a lot of attention as many telephone companies and individuals are switching to/starting a VoIP service.
In this article, I am going to explain:
1) What is VoIP (Explained above and you’ll come to know it even better as you proceed with the article)
2) Difference between PSTN and VoIP
3) VoIP Process (Working of VoIP)
4) Thing you’ll require to make a VoIP call
5) Protocols involved in VoIP (Just an overview)
6) Advantages and Disadvantages of VoIP
7) Tools and Traps
VoIP simply refers to the process of transfer of voice communications over Internet Protocol.
The main difference between traditional telephone system, PSTN (Public Switched Telephone Network) and VoIP is that, in PSTN, we use circuit switching. This means, while you are talking on the telephone, connection is established via cables. It requires bunch of wires, switches and connections. So, while you are on the call, some specific lines are dedicated between you and other party. When you are using them, no one else can use them and it remains busy until you terminate the call.
In the beginning, there were copper wires. As technology improved, fiber-optic cables came into the picture and millions of calls can be made over fiber-optic, but lines still get congested on the eve of Christmas and New Year. After that, transoceanic fibers came and now, each cable can handle more than 100 million calls.
This approach is favorable for limited users. But, nowadays, population is in millions. Imagine the resources and time required for these many users. Also, congestion occurs frequently. That is why we require VoIP.
In VoIP, we use packet switching, which means, call is routed through packets. Instead of establishing the connection first and dedicate time and resources, packet switching divides the call into number of pieces, transmits them over the network, and reassembles them on the other side.
In circuit switching, resources and time are dedicated between limited the number of users, but in VoIP, many users share the same resources. This saves the time, in turn.
Also, in circuit switching, user is involved in dedicated circuit which remains static for the duration of the call. In packet switching, packets find their own shortest and efficient path to travel across the network. This is a dynamic mode of switching. Hence, many users can use the same circuit/path, without much delay or interruption.
Working of VoIP:
Human speech is made up of analogue sound waves, which can be transmitted using simple techniques. A phone on the PSTN can represent your voice as a continuous stream of voltage changes on a wire, called as a carrier signal. When this signal reaches the other end the electric signals excite a diaphragm, which produces the analogy of your voice.
In digital telephony, a voice is not transmitted via circuit. Human voice is converted to a digital form (0s and 1s) by the transmitter and then re-produced at the other end. This conversion is achieved by sampling, which is the process of accepting many instantenous measurements of analogue signal.
Human speech consists of jumble of many different tones, ranging from very low (bass) sounds of 100 Hz up to treble sounds of about 1700 Hz and higher. With harmonics and other "noises," speech includes tones of up to about 4000 Hz. Therefore, to replicate all the sounds, speech is usually sampled at about 8000 times per second.
Human voice waveform looks like this:
Speech signal is continuous in time. To convert this signal in a digital format, the waveform is measured thousands of times per second. For every voltage level, the corresponding combination of 1’s and 0s exists. This process is known as sampling.
Enough samples should be taken to ensure the exact replication of the original voice, because if you take very few samples, it may result in multiple waveforms. Too many samples provide fantastic sound quality, but it requires too much of data to be sent across the network, which can consume more bandwidth.
The Nyquist Sampling Theorem states that a signal may be perfectly reconstructed if it is sampled at a rate greater than or equal to twice the frequency of the highest frequency component of the signal.
After taking enough samples, the data is packed in packets and sent. When these packets reach the other end, the sampled data creates original waveform, which excites a diphragm.
With the PSTN systems, your device receives power from the central office, which means even if you don’t have power at your home, you still can make calls, as long as the central office itself has the power. The same bunch of wires carry both power and voice.
In case of VoIP, your computer or device should have power. Moreover, your internet connection should be up for making the call. As soon as you lost either of the above, you cannot make any VoIP.
Things you require to make a VoIP call: (See the image above)
i) Broadband enabled computer with good ISP. We do not want any time-delays or lagging with bad ISP.
ii) In order to make actual VoIP call, you need at least one of the instruments listed below:
a) USB phone
b) Broaband Analog Telephone Adapter(BATA)
c) Dial-up Telephone Adapter
d) IP Phones
e) Dial-up phones
f) Dual Broadband/Dial-up IP phone
iii) Even if you don’t have any of the above hardware, you still can make VoIP call using just a computer. There are many companies nowadays who offer such service. Just Google “VoIP” and you’ll get lots of companies. You have to choose carefully by reviewing feedback from other users and the featueres they exibit. Different companies have different rules and exibit different features.
iv) Headphones
Protocols involved in the process of VoIP:
By its name, many people say that there is just one protocol involved i.e. IP. But it’s not what it looks like. There are many protocols involved in the successful conversation between the two (or more) parties via VoIP. These are:
1) SIP - Session Initialization Protocol
2) MGCP - Media Gateway Control Protocol
3) Megaco H.248 - Gateway Control Protocol
4) RVP over IP - Remote Voice Protocol over IP
5) SAP v2 - Session Announcement Protocol
6) SDP - Session Description Protocol
7) SGCP - Simple Gateway Control Protocol
Skinny - Skinny Client Control Protocol (SCCP)9) RTSP - Real Time Streaming Protocol for media play-out control
10) RSVP - Resource Reservation Protocol
11) STUN - Simple Traversal of UDP through NAT
12) TURN - Traversal Using Relay NAT
13) ICE - Interactive Connectivity Establishment
14) TLS - Transport layer security
VoIP Advantages:-
1) Lowering monthly bill of PSTN. Even you can even make PC-2-PC calls endlessly, for free.
2) Phone number flexibility. Your phone number will not be based on the geographical location, but it will depend on your IP, which remains the same, in most cases.
3) Realtime call management
4) High Portabliliy
5) Can be topped up anytime, from anywhere
VoIP Disadvantages:-
1) Can be costly, because it needs electricity if using on computer or VoIP enabled cell phones. Moreover, it may need a hardware device too.
2) If the internet is lagging, you may get 2-3 seconds voice delay. Moreover, the other person may hear garbage or distorted voice.
3) Bad or unreliable internet connection users should not use VoIP as VoIP relies solely on the internet.
4) Traffic can be sniffed.
Free VoIP service:
Code: Select all
http://evaphone.com/Tools and Traps:
VoIP Call Sniffers:
VOIPong is a SIP, H.323, SCCP sniffer utility that can be used to detect and capture VoIP calls. With the appropriate tools, VOIPong can dump the conversation to a separate wave file. The unfortunately name vomit (Voice over Misconfigured Internet Telephones) is an unrelated precursor to VOIPong that can translate tcpdump files of Cisco IP telephone sessions.
Author
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