The document discusses the history and technology behind voice over internet protocol (VoIP). It describes how voice transmission evolved from analog to digital formats, allowing calls to be transmitted over data networks rather than traditional circuit-switched telephone networks. Key steps included sampling analog voice using pulse code modulation (PCM) at 64 kbps, compressing the digital signal, adding headers, and transmitting over IP networks using protocols like UDP and RTP. This allows voice calls to be made reliably over the internet instead of using circuit-switched telephone lines.

1. VOICE OVER IP (VOIP)
1

2. The first voice transmission, sent by Alexander
Graham Bell, was accomplished on 10th March, 1876
 Ring Down Circuit –> No ringing involved
 “Mr. Watson -- come here -- I want to see you”
Gradually evolved from a one-way voice transmission
to a bi-directional voice transmission
 Moving the voices across the wire required a carbon
microphone, a battery, an electromagnet, and an iron
diaphragm.
 required a physical cable between each location that the
user wanted to call.
2

3. 3

4. Place a physical cable between every
household requiring access to a telephone
Such a setup is neither cost-effective nor
feasible
For N phones require
N(N-1)/2 cables
For 10 phones require
45 cables
4

5. It could map any phone to another phone
 With this device, the telephone users needed only one
cable to the centralized switch office, instead of seven
At first, a telephone operator acted as the switch
 This operator asked callers where they wanted to dial
and then manually connected the two voice paths
For N phones require N cables
After 100 years or so, the human
switch was replaced by electronic
switches
5

6. Manual control—Switch/cord boards
Patch Cord
Pairs
Off-Hook Indicator
Manual Ring
Tip
Ring
6

7. Circuit - Switching telephone network
Carries analog voice
Constant Bandwith
7

8. 1. RealTimeVoiceTransmission
2. PointTo Point Networks
3. Bandwidth is constant
4. DataTransmission
PSTN & ISDN &TDM
Circuit – Switching Network
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10. Sounds and Human speech, are all in analog form
Until several decades ago, the telephony network was
based on an analog infrastructure as well
Analog communication is neither robust nor efficient
at recovering from line noise
Telephone lines are very sensitive to inductance or
voltage produced by nearby electric circuits and lines
In the early telephony network, analog transmission
was passed through amplifiers to boost the signal
Analog communication is a mix of time and amplitude
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12. If you were far away from the end office
switch (which provides the physical cable to
your home), an amplifier might be required to
boost the analog transmission (your voice).
Analog signals that receive line noise can
distort the analog waveform and cause
garbled reception.
This is more obvious to the listener if many
amplifiers are located between your home
and the end office switch.
12

13. Figure shows that an amplifier does not clean the
signal as it amplifies, but simply amplifies the
incoming signal, including distortion, if it exists
This process of going through several amplifiers
with one voice signal is called accumulated noise
13

14. In digital networks, line noise is less of an
issue because repeaters not only amplify the
signal, but also clean it to its original
condition
This is only possible with digital
communication because such
communication is based on 1s and 0s
Therefore when signals are repeated, a clean
sound is maintained without distortion
14

15. When the benefits of this digital
representation became evident, the
telephony network migrated to pulse code
modulation (PCM)
15

16. PCM is the most common method of
encoding an analog voice signal into a digital
stream of 1s and 0s
All sampling techniques use the Nyquist
theorem, which basically states that if you
sample at twice the highest frequency on a
voice line, you achieve good-quality voice
transmission
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17. The PCM process is as follows:
 Analog waveforms are put through a voice frequency
filter to chop out anything greater than 4000 Hz
 These frequencies are filtered to 4000 Hz to limit the
amount of crosstalk in the voice network
 Using the Nyquist theorem, you need to sample at
8000 samples per second to achieve good-quality
voice transmission
 The filtered analog signal is then sampled at a rate of
8000 times per second
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19. Two basic variations of 64 kbps PCM are
commonly used:
 µ-law, the standard used in North America; and
 a-law, the standard used in Europe
The methods are similar as both use logarithmic
compression to achieve from 12 to 13 bits of
linear PCM quality in only eight-bit words, but
they differ in relatively minor details
 E.g.The µ-law method has a slight advantage over
the a-law method in terms of low-level signal-to-noise
ratio performance
19

20. Switching
Office
CPE CPE
Transmission Facilities
Switching
Office
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21. 21
1
10
9 8
7
3
2
4
5
6
1 2 3
1 2 3
1 2 3
65 66 67
228 229 230
1298 1299 1300
1 2 3 4 5
19,000
200 million telephones
19,000 end
offices
1300 toll
offices
230 primary
offices
67 sectional
offices
10 regional
offices
(full mesh)
Source: Computer Networks, Andrew S.
Class
5
Class
4
Class
3
Class
2
Class
1

22. 22

23. VoIP is the ability to make telephone calls and
send faxes over IP-based data networks with
a suitable quality of service and superior
cost/benefit.
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26. 26

27. Analog
Digital Voice
Compression to less than 32Kbps
Transfers through Routers, LAN Switches
etc, using their Protocols
27

28. 28

29. Analog
Digital
Voice
CODEC: Analog to Digital
Compress
Create Voice Datagram
Add Header
(RTP, UDP, IP, etc)
Network
29

30. Digital
Analog
Process Header
Re-sequence and
Buffer Delay
Decompress
CODEC: Digital to Analog
Network
Voice
30

31. Redundant routes through
network
Redundancy within each network
element
How Reliability Achieved
In separate gateway controllers
Mostly integrated in switching
system
Call Processing Intelligence
ATM, FR, native IP in access; ATM
native IP in core
TDM in access, edge, coreTransport
SIP, H.323DTMF, SS7Signaling
Variable 8 – 32 kb/s64 kb/sBandwidth per call
Gateways, gateway controllers,
routers
Class 4, Class 5 switching systemsNetwork Elements
NoYes
Network Resource Reserved at Call
Setup
NoYesQoS Guarantees
Packet switchingTDM circuit switchingUnderlyingTechnology
VoIPToday’s PSTN
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34. G.711 is the international standard for
encoding telephone audio on a 64 kbps
channel. It is a pulse code modulation (PCM)
scheme operating at a 8 kHz sample rate,
with 8 bits per sample, fully meeting ITU-T
recommendations.The module is designed
and tested on theTITMS320C54x platform
but can be ported to other DSP and RISC
platforms, as well as MSWindows.
Information from http://www.spiritcorp.com/
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