Introduction to

Lesson 1
Telephony & VoIP

2012

1
 Lesson Objectives

After completing this lesson, you will be able to:

• Describe how digital signaling differs from analog signaling
• Explain the basic concept of voice over IP communications
• Describe the purpose of the gateway in a VoIP network

2
The Analog Circuit

3
 Typical Analog Circuit

• The twisted pair cooper wire from the central switch office to a
subscriber's home is called a subscriber loop
• The subscriber loop handles two types of information on the same
twisted pair:
• Signaling
• Voice

Subscriber Central Office

Telephone Set Switch

4
Loop Start Signaling

Transmitter Switch hook Ringer Ring Battery and

Current
Generator Detector

Loop

Receiver

5
 On-Hook
• In on-hook stage the switch is open and there is no current flow

Telephone Switch

Local Loop Local Loop

6
 Off-Hook
• When the handset is picked up (going off-hook) a switch on the phone closes
the connection between the two wires and a -48 VDC current is drawn from the
central office switch
• The switch determines that current is being drawn and provides dial tone so the
person on the phone knows it is time to dial a number

Telephone Switch

Dial Tone

Local Loop Local Loop

7
 Dialing
• Upon hearing the dial tone, the user pushes the number buttons,
which are connected to a tone generator inside the dial, which
generates DTMF tones
• The Telephone Switch collects the DTMF digits and maps them to
a physical subscriber

Telephone Switch
Dialing
(DTMF Transmissions)

Local Loop Local Loop

8
 DTMF – Dual Tone Multi-Frequency
• DTMF is the common method of sending dialing information
(replaced pulse dialing of the original telephone networks)
• Each number is represented by two tones which are transmitted
simultaneously over the voice path
• Each row representing a low frequency and each column
representing a high frequency 1209 1336 1477 1633

697 1 2 3 A

770 4 5 6 B

852 7 8 9 C

941 * 0 # D

9
 Call Progress Tones
• In Telephony, call progress tones are audible tones sent from the
PSTN or a PBX to calling/called parties to indicate the status of
phone calls

Call Progress Tone Description

Indicates that the telephone exchange is working, has recognized

Dial Tone an off-hook, and is ready to accept digits

This tone assures the calling party that a ringing signal is being sent
Ringback Tone on the called party's line

Busy Tone Indicates to the calling party that the remote phone is occupied

Reorder Tone Indicate that a person has dialed an invalid code, or that all trunks
(Fast Busy) are busy and/or their call is unroutable

10
 Ringing

• The Telephone Switch applies an AC ringing voltage which causes the

sound mechanism of the Called Telephone to ring
• The Telephone Switch also plays a Ringback tone to assure the calling
party that a ringing signal is being sent on the called party's line

Telephone Switch

Ringback Tone Ringing Voltage

Local Loop Local Loop

11
 Conversation

• The transmitter (handset microphone) puts out an electric current

which varies in response to the acoustic pressure waves produced
by the voice
• The resulting variations in electric current are transmitted along
the telephone line to the other phone
Telephone Switch

Voice Voice

Local Loop Local Loop

12
 Call Tear Down

• When a party "hangs up" (puts the handset on the cradle), DC

current ceases to flow in the line, thus signaling to the telephone
switch to disconnect the call
• The switch plays a fast busy tone to the remote party

Telephone Switch

Fast Busy Tone

Local Loop Local Loop

13
Telephony Network (1)

415-577-3800

415-577-3801

Telephone Switch
415-577-3700
(Central Office)
415-577-3701

415-577-3722

415-577-3733

Company X
415-577-3760

415-577-3785

14
Telephony Network (2)

415-577-3800

415-577-3801

Telephone Switch
415-577-3700

415-577-3701

415-577-3702

415-577-37xx
415-577-3703
PBX Digital Trunk

415-577-3704

415-577-3705

15
Digital Communication

16
Digital Communication

• A digital trunk is a single communication path between two switches that is

used to carry many simultaneous voice conversations

Local Central Office Remote Central Office

17
 Pulse Code Modulation (PCM)
• A method of encoding an audio signal in digital format
• A standard audio signal is encoded as 8000 analog samples per
second, of 8 bits each, giving a 64 Kbit/s digital signal known as DS0
• The default signal compression encoding on a DS0 is either μ-law
(North America and Japan) or A-law (Europe and most of the rest of
the world)

18
 Time Division Multiplexing (TDM)

• Uses time-division multiplexing

1 1
64 Kbps

2 2
64 Kbps
3 2 1
3 3
64 Kbps

. . . 32 . . . 32
64 Kbps

19
 E1
• Data rate of 2.048 Mbit/s (full duplex)
• Split into 32 time slots
• Each time slot sends and receives an 8-bit sample 8000 times per second
(8 x 8000 x 32 = 2,048 Mbit/s)
• Ideal for voice telephone calls where the voice is sampled into an 8 bit number
(PCM)
• One timeslot (TS0) is reserved for framing purposes
• One timeslot (TS16) is often reserved for signaling purposes
1 2 15 17 30
Maintenance

Signaling
Framing

Voice

Voice

Voice

Voice

Voice
... ...
and

0 1 2 15 16 17 31

2.048 Mb/s

20
 T1
• Data rate of 1.544 Mbit/s
• Split into 24 time slots each encoded in 64 Kbit/s streams
• 8 Kbit/s of framing information for synchronization
• 64,000 x 24 + 8 = 1544 Mbit/s
• Timeslot (TS24) is often reserved for signaling purposes
1 2

Signaling
R
Voice
Voice

A
M
I
N
G

Frames
1.536Mb/s

21
 Signaling Methods
• In-band signaling is the exchange of signaling (call control) information on
the same B-channel that the telephone call itself is using
• CAS (Channel Associated Signaling)
Voice + Signaling Link

• Out-of-band signaling is the exchange of signaling that is done on a

channel that is dedicated for this purpose and separate from the channels
used for the telephone call
• Common Channel Signaling (CCS) such as ISDN and SS7

Signaling Link

Voice Link

22
 ISDN
• Integrated Services Digital Network is an ITU-T term for
integrated transmission of voice, video and data on the digital
public telecommunications network

• Two interfaces are available:

• PRI (Primary Rate Interface) primarily used to link PBXs and to connect a
PBX to the PSTN. Composed of 23 or 30 B-channels and one D-channel,
all at 64 Kbps
• BRI (Basic Rate Interface) an ISDN interface typically used by smaller
sites and customers. Consists of a single 16 Kbps D-channel plus 2 B-
channels for voice and/or data

23
ISDN (Q.931) Call Flow
Calling Party Called Party

ISDN Digital Trunk

Off hook, Dial Tone, Dialing

Setup

Call Proceeding

Ringback Tone Alerting

Ringing

Connect
Off hook

Voice Channel

On hook Disconnect

Release

Release complete

24
 BRI
• Point to Point

U-Interface S/T Interface

PBX
NT1

ISDN Switch

• Point to Multi-Point

U-Interface S/T Interface

NT1

TE TE TE TE
ISDN Switch

25
 BRI (cont.)

• ISDN Basic Rate Interface (BRI) service offers two B-channels and one
D-channel (2B+D)
• B-channel service operates at 64 kbps and is meant to carry user data
• D-channel service operates at 16 kbps and is meant to carry control
and signaling information
1 2

Signaling
Framing

Voice
Voice
and

0 1 2

144 Kb/s

26
Clock Synchronization
Master Clock

Timing
Toll Center

Timing Timing

End Office End Office

Timing Timing
PBX PBX

27
Voice over IP (VoIP)

28
 What is VoIP ?

• Voice over Internet Protocol (VoIP) is a set of technologies that

enable the transmission of voice traffic over IP-based networks
instead of the Public Switched Telephony Network (PSTN)

29
 Circuit vs. Packet Switching

• Circuit Switching
• Traditional voice calls, running over the PSTN, are made using circuit switching,
where a dedicated circuit or channel is set up between two points before the
users talk to one another

• Packet Switching
• Data transmission technique in which data is separated into small 'packets',
each with its own routing information and then sent through a shared, often
public, network; at the other end the packets are reassembled into the original
data format
• In this method bandwidth is only used when something is actually being
transmitted

30
 VoIP Protocol Stack

• VoIP is composed of two key components:

• Bearer (actual voice being sent over the network) using RTP/RTCP protocols

• Signaling (additional messaging that is necessary to control, establish, and tear-

down the voice calls)
The most common signaling protocols are:
• SIP
• H.323
• MGCP
• MEGACO

31
 RTP
• RTP (Real-Time Transport Protocol) is used to encapsulate VoIP
data packets inside UDP packets
• RTP provides end-to-end network transport functions suitable for
applications transmitting real-time data

V P X CC M PT Sequence Number

Time Stamp RTP Header

Synchronization Source ID - SSRC
12 octets

Voice Bits

32
Voice Codecs
• A codec (Coder/Decoder) converts analog signals to a digital
bitstream, and back into an analog signal for transmission across IP
networks
• Codecs generally provide a compression capability to save network
bandwidth
• Some codecs also support silence suppression, where silence is not
encoded or transmitted
Codec Bit Rate (kbps)
G.711 PCM (A-Law / Mu-Law) 64
G.726 ADPCM 16, 24, 32 and 40
G.729 CS-ACLEP 8
G.723.1 CELP 6.3 and 5.3

33
 VoIP Challenges

• Delay – Each component in the voice path adds delay (sender,

network, receiver). ITU-T G.114 recommends 150 msec as maximum
desired delay to achieve high voice quality
• Jitter – Variation in delay; the effects of jitter can be mitigated by
storing voice packets in a jitter buffer upon arrival and before
producing audio
• Packet loss – Occurs either in bursts or due to congested network.
Periodic loss in excess of 5-10% of all VoIP packets can degrade voice
quality significantly

34
Delay

Start Talk Sender Receiver

Network

Packet X Packet X Arrive Start Hear

Transmitted

Network Transit
Processing Delay
Processing t
Delay Delay
End-to-End Delay

35
Jitter
• Jitter (delay variation) caused when voice packets suffer different
transit delays, causing variation in arrival times at the receiver end
• The jitter buffer collects voice packets, stores them and sends them
to the voice processor in evenly spaced intervals

A B C Sender

A B C Receives

D1 D2 = D1 D3 = D2 t

36
VoIP Gateways

37
Enterprise PSTN & Data Network
Headquarters Branch

Backup
PSTN

IP

Telecommuter

38
 FXS Gateways
• FXS (Foreign Exchange Station) – Emulates a PSTN/PBX
Provides battery power, sends dial tone and generates ringing voltage
• A standard telephone/fax machine plugs into such an interface to
receive telephone services
• FXS gateways convert (in real time) loop start signaling to SIP and
variable electric current to RTP

IP Phone
IP Signaling

Local Loop IP
IP Voice

39
 FXO Gateways
• FXO (Foreign Exchange Office) – Generates the on-hook and off-
hook indicators used to signal a loop closure at the FXS end of the
circuit
• FXO gateways convert (in real time) loop start signaling to SIP and
variable electric current to RTP

PBX

IP Signaling
IP Phone
IP Voice
Local Loop
IP

40
Digital Gateway

PBX

E1 / T1

Mediant 1000 Mediant 2000

IP Signaling
E1 / T1
IP PSTN
IP Voice PCM

41
Media Processing

42