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Introduction to Telephony

Many slides courtesy of


1

Analog TelephonyPOTS Basics


Tip Ring

Sleeve

Basic Call Progress: On-Hook


Telephone Switch

Local Loop

Local Loop

-48 DC Voltage DC Open Circuit No Current Flow

Basic Call Progress: Off-Hook


Off-Hook Closed Circuit DC Current Dial Tone
Local Loop Local Loop Telephone Switch

Basic Call Progress: Dialing


Off-Hook Closed Circuit Dialed Digits Pulses or Tones
DC Current Local Loop Telephone Switch

Basic Call Progress: Switching


Off-Hook Closed Circuit
Telephone Switch Address to Port Translation

DC Current Local Loop

Local Loop

Basic Call Progress: Ringing


Off-Hook Closed Circuit
Ring Back Tone DC Current Local Loop Telephone Switch

DC Open Cct. Ringing Tone


Local Loop

Basic Call Progress: Talking


Off-Hook Closed Circuit
Voice Energy DC Current Local Loop Telephone Switch Voice Energy DC Current Local Loop

Analog TelephonySignaling

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

10

Analog Telephony Supervisory Signaling


Loop start
Current flow sensed
Switch

Ground start
Momentary ground ring lead
Switch

11

Loop Start
Station Loop (Local or Station)
Switch

PBX or Central Office

+
+

DC Current

Switch

Ringing

AC
Switch

+
12

E&M Signaling
PBXs, switches
Separate 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

13

Signaling and Addressing

Dial Pulse

DTMF

ISDN

Analog Transmission In-Band Signaling 09, *, # (12 Digits)

Digital Transmission Out-of-Band Message-Based Signaling

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Pulse Dialing
Off-Hook Make (Circuit Closed) Dialing Inter-Digit Next Digit

Break (Circuit Open)

700 ms

US:60/40 Break/Make Pulse Period (100 ms)

15

Tone Dialing
Dual Tone Multifrequency (DTMF) 1209 697 1336 1477 1633

A
Timing: 60 ms Break 40 ms Make

770

852

941

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Network Call Progress Tones


Tone
Dial
Busy Ringback, Normal

Frequency (Hz)
350 + 440
480 + 620 440 + 480

On Time
Continuous
0.5 2

Off Time

O.5 4

Ringback, PBX
Congestion (Toll) Reorder (local) Receiver Off-hook No Such Number

440 + 480
480 + 620 480 + 620

1
0.2 0.3

3
0.3 0.2 0.1

1400 + 2060 + 2450 +2600 0.1 200 to 400

Continuous, Freq. Mod 1Hz

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Voice Channel Bandwidth


Voice Channel

Output Voltage or Energy

Voice Signal

.2

Frequency (K-Hertz)

Tone Dialing Signals

Systems Control Signals


18

Local Access Network


Feeder Route Boundary

Central Office

40,000 to 50,000 Lines

Serving Area Boundary

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Switching Systems
Manual controlSwitch/cord boards
Off-Hook Indicator
Tip Ring

Patch Cord Pairs Manual Ring

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Summary

Analog voice technology dates back to the 1900s


Information exchange based on voltage, current flow, grounding, etc.

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Digital Telephony
Digital Trunking
Switch Switch

Analog Loop POTS

Digital Network
Switch

CB

Digital Loop Digital Network ISDN


Switch

22

Digital Telephony
Pulse Code ModulationNyquist Theorem
Voice Bandwidth = 300 Hz to 3400 Hz

Analog Audio Source

Sampling Stage

Codec Technique

= Sample 8 kHz (8,000 Samples/Sec)

23

Pulse Code Modulation Analog to Digital Conversion


ALaw (Europe)
Quantizing Noise

100100111011001

Stage 1

Law (USAJapan)

Quantizing Stage
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Digital TelephonyT1 and E1/J1


T1 ITU-T G.733
Sampling Frequency Channel Bit Rate Time Slots per Frame Channels per Frame Bits per Frame Framing 8 kHz DS064 kbps 24 24 24 x 8 + 1 = 193 D4/Super Frame (12) Extended Super Frame (24) Framing Indicator 193rd Bit of Frame

E1/J1 ITU-T G.732


8 kHz DS064 kbps 32 30 32 x 8 = 256 E1: Multiframe (16) J1: CRV in Bit 1 of frame 2.048 kbps Word of 7 Bits in the 0 Channel of Odd Frames 8,000 x 256 = 2.048 Mbps E1: CCS in TS 16

System Bit Rate Signaling

8,000 x 193 = 1.544 Mbps Robbed Bit Channel Associated Signaling D4/Super Frame LSB/Channel

Frame 6 and 12

Extended Super Frame CAS in TS 162 Ch Every Other Frame LSB/Channel J1: TS0 Frames 6, 12, 18, 24

36

Digital Signaling Schemes


Channel Associated Signaling
Extended Super Frame

Audio Address Signaling (DTMF)

Bit A B C D
Supervision On/Off Hook Address Signaling (Dial Pulse)

Frame 6th 12th 18th 24th

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Digital Signaling Schemes


Common Channel Signaling
E-1
Time Slot 0 Time Slot 16

Audio Address Signaling (DTMF)

Supervision On/Off Hook

Address Signaling (Dial Pulse)

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Digital Telephony Synchronization


Bit synchronization
Primary reference source

Ones density (except for J1/CMI)

Time slot synchronization


Bits/byte/channel

Frame alignment
Basic rule
193rd bit pattern
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Digital Telephony Synchronization


One Multiframe (ESF)
3 ms
1 12 24

1 Frame, 125us, 193bits 24 Time Slots


1 12 24

1 Channel Time Slot, 5.18us

648ns
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SynchronizationTraditional Network Clocking Strata


Stratum 1
Master Clock

PRS
Timing
Toll Office

.00001ppm

Timing
Timing
End Office

Timing

One per LATA

End Office

3
PBX

DCS

Bits Distribution

PBX

4
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Digital Telephony Analog Emulation and Pair Gain


Backbone to largest interoperable network in the world Signaling information exchange based on ~30 year old concepts
Twiddling bits based on ~100-year old signaling

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Digital Telephony Summary

Analog telephony emulation


Voice encoding Limited signaling Loop consolidation

32

Voice Coding and Compression


Speech coding schemes
Subjective impairment analysis: mean opinion scores Digitizing voice Voice compression
ADPCM CELP (LD-CELP and CSA-CELP)

Silence Removal Techniques (DSI using VAD)

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Voice Compression Technologies


Unacceptable
64

Business Quality
(Cellular)

PCM (G.711)

Toll Quality *

Bandwidth
(Kbps)

32 24 16 8 0

ADPCM 32 (G.726) ADPCM 24 (G.726) ADPCM 16 (G.726) LPC 4.8

LDCELP 16 (G.728) CS-ACELP 8 (G.729)

Quality
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Speech Coding Schemes


Waveform coders
Non-linear approximation of the actual waveform Examples: PCM, ADPCM

Vocoders
Synthesized voice Example: LPC

Hybrid coders
Linear waveform approximation with synthesized voice Example: CELP
35

Subjective Quality (MOS)

Subjective Impairment Analysis: Mean Opinion Scores


5 4 3
Hybrid Coders Waveform Coders

2
1 2
Score 5 4 3 2 1

Vocoders

Kbps

16

32

64

Quality Excellent Good Fair Poor Bad

Description of Impairment Imperceptible Just Perceptible, not Annoying Perceptible and Slightly Annoying Annoying but not Objectionable Very Annoying and Objectionable

Digitizing Voice: PCM Waveform Encoding


Nyquist Theorem: sample at twice the highest 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 = DS-0 64 Kbps

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Companding
Linear coding to digital results in small amplitude signals having poorer resolution than larger amplitude signals
u-Law and A-Law coding uses more bits for smaller amplitude signals Result: uniform SQR (signal to noise quantization ratio) across input range
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Non-Linear vs. Linear Encoding


Output

Output

Input

Input

Non- Linear Encoding


Closely Follows Human Voice Characteristics High Amplitude signals have more quantization distortion

Linear Encoding
Relatively easy to analyze, synthesize and regenerate All amplitudes have roughly equal quantization distortion

39

Waveform Coders

Quantizing

Encoding

Sampling

Filtering

1110010010010110

Waveform ENCODER

Waveform DECODER

40

Voice Compression
Objective: reduce bandwidth consumption
Compression algorithms are optimized for voice Unlike data compression: these are loose

Drawbacks/tradeoffs
Quantization distortion Tandem switching degradation Delay (echo)

41

Digital Speech Interpolation (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 the window before cells are constructed Automatically disabled for fax/modem

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Voice Activity Detection


- 31 dbm B/W Saved Voice Activity (Power Level) - 54 dbm Pink Noise Voice Spurt Time
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Hang Timer
SID Buffer

No Voice Traffic Sent SID

Silence

Voice Spurt

Bandwidth Requirements
Voice Band Traffic
Encoding/ Compression
G.711 PCM A-Law/u-Law G.726 ADPCM G.729 CS-ACELP G.728 LD-CELP G.723.1 CELP

Result Bit Rate


64 kbps (DS0)

16, 24, 32, 40 kbps 8 kbps 16 kbps 6.3/5.3 kbps Variable

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Voice Quality
Anything Above an MOS of 4.0 Is Toll Quality
Compression Method MOS Score
Delay (msec)

64K PCM (G.711)


32K ADPCM (G.726) 16K LD-CELP (G.728) 8K CS-ACELP (G.729) 8K CS-ACELP (G.729a)

4.4
4.2 4.2 4.2 3.6

0.75
1 35 15 15

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Voice Network Transport


Voice Network Transport is typically TDM circuit-based:
T1/E1

DS3/E3
SONET (OC3, OC12, etc.)

But can also be packet-based:


ATM Frame Relay

IP

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Delay
Sender
PBX

Receiver Network
PBX

First Bit Transmitted

Last Bit Received

A
Processing Delay Network Transit Delay

A
t
Processing Delay

End-to-End Delay
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Voice Transport and Delay


Cumulative Transmission Path Delay

CB Zone Satellite Quality High Quality


0 100 200 300 400 500

Fax Relay, Broadcast


600 700 800

Time (msec) Delay Target

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Delay VariationJitter
Sender
Network

Receiver

Sender Transmits

t A D1 D2 = D1 B C
Sink Receives

D3 = D2

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Adaptive Clocking
Transmit Clock
In-Bound Cells Outbound Frames

Reassembly FIFO Queue

PBX1

Transport Network

PBX2

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