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Introduction to Telephony
Many slides courtesy of
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Analog Telephony—POTS Basics
Tip
Ring
Sleeve
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Basic Call Progress: On-Hook
Telephone
Switch
Local
Loop
Local
Loop
-48 DC Voltage
DC Open Circuit
No Current Flow
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Telephone
Switch
Local
Loop
Local
Loop
Basic Call Progress: Off-Hook
DC Current
Dial Tone
Off-HookClosed
Circuit
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Basic Call Progress: Dialing
Dialed Digits
Pulses or
Tones
DC Current
Telephone
Switch
Local
Loop
Off-HookClosed
Circuit
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DC Current
Telephone
Switch
Local
Loop
Basic Call Progress: Switching
Addressto
Port
TranslationLocal
Loop
Off-HookClosed
Circuit
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Basic Call Progress: Ringing
Ring Back
Tone
DC Current
DC Open Cct.
Ringing Tone
Telephone
Switch
Local
Loop
Local
Loop
Off-HookClosed
Circuit
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Basic Call Progress: Talking
Voice Energy
DC Current
Voice Energy
DC Current
Telephone
Switch
Local
Loop
Local
Loop
Off-HookClosed
Circuit
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Analog Telephony—Signaling
• Supervisory
• Addressing
• Call progress
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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
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Analog Telephony— Supervisory Signaling
• Loop start
Current flow
sensed
• Ground start
Momentary ground
ring lead
Switch Switch
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Loop Start
Loop(Local or Station)
+
–
+
–
AC
+
–
Station PBX or Central Office
DC Current
Ringing
Switch
Switch
Switch
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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
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Signaling and Addressing
Analog Transmission
―In-Band‖ Signaling
0 –9, *, # (12 Digits)
Digital Transmission
―Out-of-Band‖
Message-BasedSignaling
ISDNDial Pulse DTMF
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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
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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
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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
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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
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Local Access Network
Central
Office
Feeder Route Boundary
40,000 to
50,000 LinesServing
Area
Boundary
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Switching Systems
• Manual control—Switch/cord boards
Patch CordPairs
Off-Hook Indicator
Manual Ring
Tip
Ring
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Summary
• Analog voice technology dates
back to the 1900’s
• Information exchange based onvoltage, current flow, grounding, etc.
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Digital Telephony
Digital TrunkingSwitchSwitch
Analog Loop Digital Network
POTSSwitch
CB
Digital Loop Digital Network
ISDNSwitch
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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
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Pulse Code Modulation— Analog to Digital Conversion
Stage 1
Quantizing Stage
Quantizing Noise
100100111011001
A—Law (Europe)
µ—Law (USA –Japan)
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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
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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)
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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
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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 alignmentBasic rule
193rd bit pattern
Digital Telephony
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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
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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
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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
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Digital Telephony Summary
• Analog telephony emulation
Voice encoding
Limited signaling
Loop consolidation
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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
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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)
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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 codersLinear waveform approximation withsynthesized voice
Example: CELP
Subjective ImpairmentA l i M O i i
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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‖
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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
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Adaptive Clocking
In-Bound CellsOutbound
Frames
Transmit Clock
Reassembly FIFO Queue
Transport
Network
PBX1 PBX2
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