Introduction to Telephony and VoIP

7/27/2019 Introduction to Telephony and VoIP

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Introduction to Telephony

and Voice over IP (VoIP)


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The Analog Circuit


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Typical Analog Circuit

The twisted pair wires from the central switch office to a subscriber's home is

called a subscriber loop

The subscriber loop handles two types of information: signals and voice on the

same twisted pair


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Loop Start Signaling


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On-Hook

In on-hook stage the switch is open and there is no current flow


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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


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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


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DTMF - Dual Tone Multi-Frequency

1 2 3 A

4 5 6 B

7 8 9 C

* 0 # D

1209 1336 1477 1633

697

770

852

941

DTMF is the common method of sending dialing information (replaced pulse

dialing)

Each number is represented by two tones which are transmitted simultaneously

on the voice path

Each row representing a lowfrequency and each column representing a high

frequency


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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


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Conversation

The transmitter (handsets 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


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Call Tear Down

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

to flow in that line, thus signaling to the telephone switch to disconnect the call

The switch plays a fast busy tone to the remote party


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Call Progress Tones

Call Progress Tone Description

Dial ToneIndicates that the telephone exchange is working, has

recognized an off-hook, and is ready to accept digits

Ringback ToneThis tone assures the calling party that a ringing signal is

being sent on the called party's line

Busy ToneIndicates to the calling party that the remote phone is

occupied

Reorder Tone (Fast

Busy)Indicate that a person has dialed an invalid code, or that all

trunks are busy and/or their call is unroutable

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.


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Telephony Network (1)

Central Office)

415-577-3800

415-577-3801

415-577-3700

415-577-3701

415-577-3722

415-577-3733

415-577-3760

415-577-3785


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Telephony Network (2)

415-577-3800

415-577-3801

415-577-3700

415-577-3701

415-577-3702

415-577-3703

415-577-3704

415-577-3705

415-577-37xx


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Digital Communication


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Digital Communication

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

to carry many simultaneous voice conversations


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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 -law (North America

and Japan) or A-law (Europe and most of the rest of the world)


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Time Division Multiplexing (TDM)

64 Kbps

64 Kbps

64 Kbps

64 Kbps

1

2

3

. . . 32

1

2

3

. . . 32

123

Uses time-division multiplexing


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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


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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


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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)

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

dedicated for the purpose and separate from the channels used for the telephone call

Common Channel Signaling (CCS) such as ISDN and SS7


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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 aPBX 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


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ISDN (Q.931) Call Flow

Voice Channel


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BRI

NT1U-Interface S/T Interface

ISDN Switch

NT1U-Interface

TE

S/T Interface

TE TE TE

ISDN Switch

Point to Point

Point to Multi-Point

PBX


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BRI (cont.)

Signaling

The 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


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Clock Synchronization

PBX

Master Clock

Toll Center

PBX

Timing

TimingTiming

End Office End Office

Timing Timing


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Voice over IP (VoIP)


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What is VoIP

VoIP is a set of technologies that enable the transmission of

voice traffic over IP-based networks instead of the Plain Old

Telephone System (POTS)


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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 beingtransmitted


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VoIP Protocol Stack

VoIP is composed of two key components:

The bearer (the actual voice being sent over the network) using the

RTP / RTCP protocols

The signaling (which are additional messaging necessary to control,

establish and tear-down the voice calls).

The most common signaling protocols are: SIP, H.323, MGCP and

MEGACO


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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

RTP Header

12 octets


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Voice Codecs

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

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


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VoIP Challenges

Delay - Each component in the 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


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Delay

Packet X

Transmitted

Network

Sender Receiver

tNetwork TransitDelayProcessingDelay

Processing

DelayEnd-to-End Delay

Start Talk

Packet X

ArriveStart Hear


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Jitter

Jitter (delay variation) caused when voice packets suffer different transitdelays, causing variation in arrival times at the receiver

The jitter buffer collects voice packets, stores them and sends them to the

voice processor in evenly spaced intervals

t

t

Sender

Receives

A B C

A B C

D1 D2 = D1 D3 = D2


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VoIP Gateways


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BranchHeadquarters

Telecommuter

PSTN

Enterprise PSTN & Data Network

IP


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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


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FXO Gateways

FXO (Foreign Exchange Office) Generates the on-hook and off-hookindicators used to signal a loop closure at the FXS's end of the circuit.

Analog telephone handsets, fax machines and (analogue) modems areFXO devices

FXO gateways convert (in real time) loop start signaling to SIP and variableelectric current to RTP


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Voice

FXS Call Flow

IP

Dial Tone

DialingINVITE

100 Trying

180 RingingRingback Tone

200 OK

ACK

BYE

200 OK


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Digital Gateway

IP

Mediant 2000

PBX

Mediant 1000

PSTNE1 / T1

E1 / T1

PCM

Digital gateways convert (in real time) ISDN or CAS signaling to SIP andPCM to RTP


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Voice

ISDN Call Flow

IP

Setup

Call Proceeding

INVITE

100 Trying

180 Ringing

Alert

200 OK

ACK

BYE

200 OK

Connect

Disconnect

Release Complete

Release


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Voice

T1 CAS (Wink Start) Call Flow

IP

Line Seizure

Wink

INVITE

100 Trying

180 Ringing

Wink Optional

200 OK

ACK

ANI

Prefix/Authorization Code/

Called Number

Ringback Tone

Answer Supervision


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Media Processing

Digital Audio Source (PCM)

Analog Audio Source

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Introduction to Telephony and VoIP

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