CISCO Introduction to Telephony

8/23/2019 CISCO Introduction to Telephony

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1

Introduction to Telephony

Many slides courtesy of



2

Analog Telephony—POTS Basics

Tip

Ring

Sleeve



3

Basic Call Progress: On-Hook

Telephone

Switch

Local

Loop

Local

Loop

-48 DC Voltage

DC Open Circuit

No Current Flow



Telephone

Switch

Local

Loop

Local

Loop

Basic Call Progress: Off-Hook

DC Current

Dial Tone

Off-HookClosed

Circuit



Basic Call Progress: Dialing

Dialed Digits

Pulses or

Tones

DC Current

Telephone

Switch

Local

Loop

Off-HookClosed

Circuit



DC Current

Telephone

Switch

Local

Loop

Basic Call Progress: Switching

Addressto

Port

TranslationLocal

Loop

Off-HookClosed

Circuit



Basic Call Progress: Ringing

Ring Back

Tone

DC Current

DC Open Cct.

Ringing Tone

Telephone

Switch

Local

Loop

Local

Loop

Off-HookClosed

Circuit



Basic Call Progress: Talking

Voice Energy

DC Current

Voice Energy

DC Current

Telephone

Switch

Local

Loop

Local

Loop

Off-HookClosed

Circuit



Analog Telephony—Signaling

• Supervisory

• Addressing

• Call progress



Off-Hook Signaling

• Loop Start (almost all telephones)

Seizure is detected when current flows

through local loop, due to off-hook

• Ground Start (PBXs)

Seizure is detected when one wire

is grounded

Seizure can be initiated in both directions



Analog Telephony— Supervisory Signaling

• Loop start

Current flow

sensed

• Ground start

Momentary ground

ring lead

Switch Switch



Loop Start

Loop(Local or Station)

+

–

+

–

AC

+

–

Station PBX or Central Office

DC Current

Ringing

Switch

Switch

Switch



13

E&M Signaling

• PBXs, switchesSeparate signaling leads for each direction

E-Lead (inbound direction)

M-Lead (outbound direction)

Allows independent signaling

State

On-Hook

Off-Hook

E-Lead

Open

Ground

M-Lead

Ground

Battery Voltage



14

Signaling and Addressing

Analog Transmission

―In-Band‖ Signaling

0 –9, *, # (12 Digits)

Digital Transmission

―Out-of-Band‖

Message-BasedSignaling

ISDNDial Pulse DTMF



15

Pulse Dialing

Make

(Circuit Closed)

Break

(Circuit Open)

Off-Hook Dialing Inter-Digit Next Digit

Pulse Period

(100 ms)

US:60/40 Break/Make

700 ms



16

Tone Dialing

1 2 3 A

4 5 6 B

7 8 9 C

* 0 # D

Dual Tone Multifrequency (DTMF)

1209 1336 1477 1633

697

770

852

941

Timing:

60 ms Break

40 ms Make



17

Network Call Progress Tones

Tone Frequency (Hz) On Time Off Time

Dial

Busy

Ringback, Normal

Ringback, PBX

Congestion (Toll)

Reorder (local)

Receiver Off-hook

No Such Number

350 + 440

480 + 620

440 + 480

440 + 480

480 + 620

480 + 620

1400 + 2060 + 2450 +2600

200 to 400

Continuous

0.5 O.5

2 4

1 3

0.2 0.3

0.3 0.2

0.1 0.1

Continuous, Freq. Mod 1Hz



18

Voice Channel Bandwidth

Output

Voltage

or

Energy

Frequency

(K-Hertz)1 2 3 4.2

Voice Signal

Voice Channel

Tone Dialing

Signals

Systems Control

Signals



19

Local Access Network

Central

Office

Feeder Route Boundary

40,000 to

50,000 LinesServing

Area

Boundary



20

Switching Systems

• Manual control—Switch/cord boards

Patch CordPairs

Off-Hook Indicator

Manual Ring

Tip

Ring



21

Summary

• Analog voice technology dates

back to the 1900’s

• Information exchange based onvoltage, current flow, grounding, etc.



22

Digital Telephony

Digital TrunkingSwitchSwitch

Analog Loop Digital Network

POTSSwitch

CB

Digital Loop Digital Network

ISDNSwitch



23

Digital Telephony

= Sample

8 kHz (8,000 Samples/Sec)Codec Technique

Sampling StageAnalog Audio Source

Pulse Code Modulation—Nyquist Theorem

Voice Bandwidth =

300 Hz to 3400 Hz

P l C d M d l ti



24

Pulse Code Modulation— Analog to Digital Conversion

Stage 1

Quantizing Stage

Quantizing Noise

100100111011001

A—Law (Europe)

µ—Law (USA –Japan)



Digital Telephony—T1 and E1/J1T1 ITU-T G.733 E1/J1 ITU-T G.732

Sampling Frequency

Channel Bit Rate

Time Slots per Frame

Channels per Frame

Bits per Frame

System Bit Rate

Framing

Signaling

8 kHz 8 kHz

DS0—64 kbps DS0—64 kbps

24 32

24 30

24 x 8 + 1 = 193 32 x 8 = 256

D4/Super Frame (12)

Extended Super Frame (24)

E1: Multiframe (16)

J1: CRV in Bit 1 of frame

Framing Indicator 193rd Bit of Frame 2.048 kbps Word of 7 Bits in the 0 Channelof Odd Frames

8,000 x 193 = 1.544 Mbps 8,000 x 256 = 2.048 Mbps

E1: CCS in TS 16

CAS in TS 16—2 ChEvery Other Frame

J1: TS0

―Robbed Bit‖ Channel Associated Signaling

D4/Super Frame Extended Super Frame

LSB/Channel LSB/Channel

Frame 6 and 12 Frames 6, 12, 18, 24

36



26

Digital Signaling Schemes

Channel Associated Signaling

Bit Frame

A 6th

B 12th

C 18th

D 24th

Extended Super Frame

Supervision

On/Off Hook

Address Signaling

(Dial Pulse)

Audio

Address Signaling

(DTMF)



27

Supervision

On/Off Hook

Address Signaling

(Dial Pulse)

Digital Signaling Schemes

Common Channel SignalingTime Slot 16

Audio

Address Signaling

(DTMF)

E-1Time Slot 0

Digital Telephony



28

Digital Telephony— Synchronization

• Bit synchronization

Primary reference source

Ones density (except for J1/CMI)

• Time slot synchronization

Bits/byte/channel

• Frame alignmentBasic rule

193rd bit pattern

Digital Telephony



29

Digital Telephony— Synchronization

3 ms

One Multiframe (ESF)

1 24

1 24

12

1 Frame,

125us, 193bits 24 Time Slots

12

1 Channel TimeSlot, 5.18us

648ns

Synchronization Traditional



30

Synchronization—TraditionalNetwork Clocking Strata

PBX

PRS

Master Clock

Toll Office

PBX

Stratum

1

2

3

4

Timing

TimingTiming

Timing

One per LATA

Bits Distribution

.00001ppm

End Office

DCS

End Office

Digital Telephony



31

Digital Telephony— Analog Emulation and Pair Gain

• Backbone to largestinteroperable network

in the world• Signaling information

exchange based on~30 year old concepts

Twiddling bits based on~100-year old signaling



32

Digital Telephony Summary

• Analog telephony emulation

Voice encoding

Limited signaling

Loop consolidation



33

Voice Coding and Compression

• Speech coding schemes

• Subjective impairment analysis: meanopinion scores

• Digitizing voice

• Voice compression

ADPCM

CELP (LD-CELP and CSA-CELP)

Silence Removal Techniques (DSI using VAD)

Voice Compression



34

Voice CompressionTechnologies

Bandwidth

(Kbps)

Quality

Unacceptable Business

Quality

Toll

Quality

8

16

32

24

64

0

*PCM (G.711)

*ADPCM 32 (G.726)

*ADPCM 24 (G.726)

*ADPCM 16 (G.726)

*LDCELP 16 (G.728)

*CS-ACELP 8 (G.729)

*LPC 4.8

(Cellular)



35

Speech Coding Schemes

• Waveform coders

Non-linear approximation of the actual waveform

Examples: PCM, ADPCM

• Vocoders

Synthesized voice

Example: LPC

• Hybrid codersLinear waveform approximation withsynthesized voice

Example: CELP

Subjective ImpairmentA l i M O i i



j pAnalysis: Mean OpinionScores

5

4

3

2

1

2 4 8 16 32 64Kbps

S u b j e c t i v

e Q u a l i t y ( M O S

)

Hybrid Coders

Vocoders

Waveform Coders

Score Quality Description of Impairment

5

4

3

2

1

Excellent

Good

Fair

Poor

Bad

Imperceptible

Just Perceptible, not Annoying

Perceptible and Slightly Annoying

Annoying but not Objectionable

Very Annoying and Objectionable

Digitizing Voice: PCM



37

Digitizing Voice: PCMWaveform Encoding

• Nyquist Theorem: sample at twice thehighest frequency

Voice frequency range: 300-3400 Hz

Sampling frequency = 8000/sec (every 125us)

Bit rate: (2 x 4 Khz) x 8 bits per sample

= 64,000 bits per second (DS-0)

• By far the most commonly used method

CODEC

PCM

64 Kbps= DS-0

C di



38

Companding

• Linear coding to digital results insmall amplitude signals havingpoorer resolution than larger amplitude signals

• u-Law and A-Law coding uses morebits for smaller amplitude signals

• Result: uniform SQR (signal to noisequantization ratio) across input range

Non-Linear vs. Linear



39

Non Linear vs. Linear Encoding

Non- Linear EncodingClosely Follows Human Voice Characteristics

High Amplitude signals have more quantization distortion

Input

Output

Linear EncodingRelatively easy to analyze, synthesize and regenerate

All amplitudes have roughly equal quantization distortion

Input

Output

W f C d



40

Waveform Coders

S a

m p l i n g

F

i l t e r i n g

E n

c o d i n g

Q u a n t i z i n g

1110010010010110

Waveform

ENCODER

Waveform

DECODER

V i C i



41

Voice Compression

• Objective: reduce bandwidthconsumption

Compression algorithms are optimized for voice

Unlike data compression: these are ―loose‖

• Drawbacks/tradeoffs

Quantization distortion

Tandem switching degradation

Delay (echo)

Digital Speech Interpolation



42

g p p(DSI)

• Voice Activity Detection (VAD)

• Removal of voice silence

• Examines voice for power, change of power,frequency and change of frequency

• All factors must indicate voice ―fits into thewindow‖ before cells are constructed

• Automatically disabled for fax/modem

Voice Activity Detection



43

Voice Activity Detection

Voice ―Spurt‖ Silence

Pink Noise

Time

Voice

Activity(Power

Level)SID Buffer SID

Hang Timer No VoiceTraffic Sent

B/W Saved

- 54 dbm

- 31 dbm

Voice ―Spurt‖

Bandwidth Requirements



44

Bandwidth Requirements

Voice Band TrafficEncoding/

Compression

Result

Bit Rate

G.711 PCMA-Law/u-Law 64 kbps (DS0)

G.726 ADPCM 16, 24, 32, 40 kbps

G.729 CS-ACELP 8 kbps

G.728 LD-CELP 16 kbps

G.723.1 CELP 6.3/5.3 kbps

Variable

Voice Quality



45

Voice Quality

Compression Method MOS Score Delay

(msec)

64K PCM (G.711) 4.4 0.75

32K ADPCM (G.726) 4.2 1

16K LD-CELP (G.728)

8K CS-ACELP (G.729) 4.2 15

8K CS-ACELP (G.729a) 15

3 –54.2

3.6

Anything Above an MOS of 4.0 Is ―Toll‖ Quality

Voice Network Transport



46

Voice Network Transport

• Voice Network Transport is typicallyTDM circuit-based:

T1/E1

DS3/E3

SONET (OC3, OC12, etc.)

• But can also be packet-based:

ATM

Frame Relay

IP

Delay



47

Delay

First BitTransmitted

Last BitReceived

Network

A A

Sender Receiver

t

PBX PBX

Network

Transit

DelayProcessing

Delay

Processing

Delay

End-to-End Delay

Voice Transport and Delay



48

Voice Transport and Delay

Time (msec)

0 100 200 300 400

CB Zone

Cumulative Transmission Path Delay

Satellite Quality

Fax Relay, BroadcastHigh Quality

Delay Target

500 600 700 800

Delay Variation—―Jitter‖



49

Delay Variation— Jitter

t

t

Sender Transmits

Sink Receives

A B C

A B C

D1 D2 = D1

Sender Receiver

Network

D3 = D2

Adaptive Clocking



50

Adaptive Clocking

In-Bound CellsOutbound

Frames

Transmit Clock

Reassembly FIFO Queue

Transport

Network

PBX1 PBX2

CISCO Introduction to Telephony

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