Introduction to Voice technologies
1
Voice over IP introduction • VoIP = Voice + IP • VOICE
Traditionally, voice was transmitted using a separate dedicated infrastructure and it is still in place i.e. PSTN The first network that was put in place was for voice ONLY.
Based on TDM
2
Voice over IP introduction (contd..)
• VoIP = Voice + IP
• TCP/IP based Data Networks
Most common data network implementations are based on TCP/IP. Internet and most business networks are also based on TCP/IP. The purpose of data networks is to transfer & share computer data between users
3
Voice & Data Network infrastructure
• VOICE
Circuit Switching
Phones/terminals Signaling Routing Transmission facilities
• DATA
Packet Switching
Data Terminals Signaling Routing Transmission facilities
4
What is meant by Data?
• Computer Data
• Voice • Video • What is common in all of them?
They can all be represented as bits i.e. these are all different forms of information As all can be represented as digital data making Voice/Video/Data integration possible
5
Voice technologies
• Voice in PSTN (TDM Based) • Voice over Packet (VoIP, VoFR or VoATM)
6
Voice over IP (contd..)
• Transport voice traffic using IP • Voice over the Internet?
Interconnected networks Applications: e-mail, file transfer, e-com
• The greatest challenges
Voice quality and bandwidth
Control and prioritize the access
• Internet: best-effort transfer
The next generation VoIP != Internet telephony
7
IP (Internet Protocol)
• A packet-based protocol
Routing on a packet-by-packet base
• Packet transfer with no guarantees
May not receive in order May be lost or severely delayed
• TCP/IP
Retransmission Assemble the packets in order Congestion control Useful for file-transfers and e-mail
8
Voice over IP Protocols
Presentation Session Transport Network Link Physical
G.729(A)/G.723(.1)/G.711 H.323/MGCP/SIP RTP/UDP/RSVP IP/WFQ/IP-prec MLPPP/FR/ATM AAL1 –––
9
Why VoIP?
• Why carry voice?
Internet supports instant access to anything “Dot-com” Many new services and applications However, voice services provide more revenues
• Why use IP for voice?
Circuit-switching is not for datacom IP-based Packet switching: Equipment cost, integrated access, less bandwidth, and widespread availability
10
Lower Equipment Cost
– PSTN switch Proprietary – hardware, OS, applications High operation and management cost Training, support and feature development cost – Mainframe computer – The IP world Standard hardware and mass-produced Application software is quite separate – IN does not match the openness and flexibility of IP A few highly successful services
11
Voice/Data Integration
– Click to talk application Personal communication E-commerce CTI – Computer Telephony Integration – Web collaboration Shop on-line with a friend at another location
– Video conferencing
– IP-based PBX – IP-based call centers
12
Enterprise Voice Over IP Applications
• Toll bypass
Most common application
• PBX extension
Saves costs by reducing maintenance costs and overhead
• H.323 interoperability
Supports voice-enabled Web applications
13
Cisco ―Voice over‖ Applications
14
Connection Types
• Local
• On-net • Off-net • PLAR • PBX-to-PBX
• On-net to Off-net
15
Local Connections
555-4001
Between two FXS Stations
555-4002
16
On-net Connections
Site A
Site B
IP Router Gateway Router Gateway
Calls within an enterprise
17
On-net Connections (contd..)
Branch A
192.168.1.1 192.168.1.254 172.16.1.254
Branch B
Soft Phone
Internet
IP Phone
18
Off-net Connections
Dial Access code: 9 Then PSTN number Branch A
192.168.1.1 192.168.1.254 172.16.1.254
Branch B
FR/ATM
PSTN
19
Tie Line Trunks
PBX
PBX
IP,FR ATM
Router Gateway
Router Gateway
20
On to off-net Connections
Branch A
192.168.1.1 192.168.1.254
PSTN
Branch B
172.16.1.254
Internet
21
Toll Bypass Using 3600
PBX
PBX
PSTN
3620
V
QoS WAN
(Intranet)
3640
4 to 12 Analog ports
V
Branch Office
Headquarters
22
Introduction to PSTN
Legacy Voice Infrastructure
23
Addressing in Telephone Systems
• Numbering is never flat, it is always hierarchical • E.163 Standard (replaced by E.164)
• E.164 ITU-T standard for ISDN numbers
• In switching terminology the numbers are termed as DNs or (Directory Numbers)
24
Dialing Types
• Pulse
Each digit is represented as a series of pulses.
• Touch Tone (DTMF)
Each digit represented as a pair of frequencies
25
Pulse Dialing Scheme
Make = Circuit Closed
Off-Hook
Dialing Inter-Digit Delay Next Digit
Break = Open Circuit 700 ms
Pulse Period (100 ms)
Supported on Cisco routers
26
DTMF Dialing
Supported on Cisco routers
Dual Tone Multifrequency (DTMF) 1209 1336 1477 1633
697
1
2
3
A
770
4
5
6
B
852
7
8
9
C
941
*
0
#
D
27
Types of circuit switched calls
Call on same switch a calling b
Call established through multiple switches c calling d
End-Office Central Office Local Exchange CLASS 5 switch
Tandem for calls within city Transit for calls out of city
28
Introduction to Signaling
The main purpose of Signaling is to setup and tear down a call and providing supervisory functions.
Signaling Classification
Off-hook Dial-tone Ringing Busy Tone Hookflash ISDN Q.931
Subscriber Signaling
Trunk or Inter-switch SS1-6 SS7 Signaling Router-Router R2 (Analog / PCM H.323 / SIP MGCP
29
Types of Signaling
Method of communicating telephony events: Off-hook, busy, on-hook…
Analog
• 2-wire • Loop start • Ground start • • • • E&M 2-wire, 4-wire Five types I-V (Cisco I,II,III,V)
Digital
• Digital subscriber lines: 2-wire, 4-wire • Digital trunks: 4-wire • Channel associated signaling (CAS) • In-band signaling
• Common channel signaling (CCS) • Out-of-band signaling
30
Basic Local Call Flow
31
Subscriber signaling for local calls
32
Basic Call Progress: On-Hook
Telephone Switch
Local Loop
Local Loop
-48 DC Voltage DC Open Circuit No Current Flow
33
Basic Call Progress: Off-Hook
Off-Hook Closed Circuit DC Current Dial Tone Local Loop Local Loop
Telephone Switch
34
Basic Call Progress: Dialing
Off-Hook Closed Circuit Dialed Digits Pulses or Tones
Telephone Switch
DC Current
Local Loop
35
Basic Call Progress: Switching
Off-Hook Closed Circuit Telephone Switch Address to Port Translation
DC Current
Local Loop
Local Loop
36
Basic Call Progress: Ringing
Off-Hook Closed Circuit Ring Back Tone DC Current Local Loop
Telephone Switch
DC Open Cct. Ringing Tone
Local Loop
37
Basic Call Progress: Talking
Off-Hook Closed Circuit Voice Energy DC Current Local Loop
Telephone Switch Voice Energy DC Current Local Loop
38
Common Terms
• Local Loop • Switches • Trunks
39
Switch Types
• Local Exchange / CO • PBX
• Tandem
• Transit
Switches solve the N² problem
40
Trunk Types
• Private Trunks
• CO Trunks • FXO Trunks • FXS Trunks • DID/DOD Trunks
• Inter-office trunks
41
2-to-4 wire conversion
• Done in Telephone Set • Done on Switch side as well
Result: ????
42
Speech-Coding Techniques
43
Introduction
• Codecs / Speech coding schemes
• Subjective impairment analysis: MOS • Digitizing voice • Voice compression
ADPCM CELP Silence Removal Techniques (DSI using VAD)
• Processing Power
A balance between quality and cost
44
Voice Quality Measure
• Bandwidth is easily quantified Voice quality is subjective • MOS, Mean Opinion Score ITU-T Recommendation P.800 Excellent – 5 Good – 4 Fair – 3 Poor – 2 Bad – 1 A minimum of 30 people Listen to voice samples or in conversations
45
ITU-T Voice Quality Standards
P.800 recommendations
The selection of participants The test environment Explanations to listeners Analysis of results
Toll quality
A MOS of 4.0 or higher
46
ITU-T Voice Quality Standards
• Subjective and objective quality-testing techniques
• PSQM – Perceptual Speech Quality Measurement
ITU-T P.861
algorithmic comparison between the output signal and a known input
type of speaker, loudness, delay, active/silence frames, clipping, environmental noise
47
Voice Compression Technologies
Unacceptable
64 (Cellular)
Business Quality
PCM (G.711)
Toll Quality *
Bandwidth
(Kbps)
32 24 16
ADPCM 32 (G.726) ADPCM 24 (G.726) ADPCM 16 (G.726) LPC 4.8
*
*
*
LDCELP 16 (G.728) CS-ACELP 8 (G.729)
*
8
0
*
*
Quality
48
Speech Waveforms & PSD
• Voiced speech
• Power spectrum density
49
Speech Waveforms & PSD (contd..)
• Unvoiced speech
• Power spectrum density
50
Type of Speech Coders
• Waveform codecs
Sample and code High-quality and not complex
Large amount of bandwidth
• Source codecs (Vocoders)
Match the incoming signal to a mathematical model Linear-predictive filter model of the vocal tract The information is sent rather than the signal Low bit rates, but sounds synthetic Higher bit rates do not improve much
51
Types of codecs
• Hybrid codecs
Attempt to provide the best of both Perform a degree of waveform matching Utilize the sound production model Quite good quality at low bit rate
52
Waveform Coders
Quantizing
Encoding
Sampling
Filtering
1110010010010110
Waveform ENCODER
Waveform DECODER
53
Vocoders
Quantizing PCM Encoder Encoding
111001001001011
PCM Decoder
Sampling
Filtering
Sample Frames
VocalCords Throat Nose Mouth
Model Parameters
10110010
Parameters
Human Speech Model
Analysis
Synthesis
54
Model Parameters
Voice Digitization
• Analog-to-Digital Conversion
discrete samples of the waveform and represent each sample by some number of bits A signal can be reconstructed if it is sampled at a minimum of twice the maximum freq.
• Human speech
0-4KHz (300-3400 Hz used in telephony) 8000 samples per second
55
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
56
G.711
• The most common codec
Used in circuit-switched telephone network PCM, Pulse-Code Modulation
• •
Uniform quantization (not done)
12 bits * 8 k/sec = 96 kbps
Non-uniform quantization
64 kbps DS0 rate mu-law North America & Japan A-law Other countries, including Pakistan A MOS (Mean Opinion Score) of about 4.3
57
DPCM
•DPCM, Differential PCM
Only transmit the difference between the predicated value and the actual value Voice changes relatively slowly It is possible to predict the value of a sample based on the values of previous samples
The receiver performs the same prediction
The simplest form • No prediction
58
ADPCM
•
ADPCM, Adaptive DPCM
Predicts sample values based on Past samples Factoring in some knowledge of how speech varies over time The error is quantized and transmitted Fewer bits required G.721 32 kbps G.726 A-law/mu-law PCM -> 16, 24, 32, 40 kbps
An MOS of about 4.0 at 32 kbps
59
CELP
• Code excited linear predictive
Hybrid coding scheme
• Very high voice quality at low bit rates, processor intensive, use of DSPs
• G.728: LD CELP—16 Kbps
Smaller Codebook
• G.729: CS ACELP—8 Kbps
G.729a variant— ―stripped down‖ 8 kbps (with a noticeable quality difference) to reduce processing load, allows two voice channels encoded per DSP
60
G.729 an Advanced CODEC
Cake
Code Excited Linear Prediction (CELP) Consumes ~ 8 Kbps
Cake Recipe $0.32 10.1.1.1
A/D 16-Bit Linear PCM
Code
DSP
Packet
Code Look-Up
• DSP = Digital Signal Processing
Ingredients:
A-sound K-sound
Directions:
Play K, A, and K
Recipe or Code Book
61
G.729x
• G.729.B
VAD, Voice Activity Detection Based on analysis of several parameters of the input The current frames plus two preceding frames DTX, Discontinuous Transmission Send nothing or send an SID frame SID frame contains information to generate comfort noise CNG, Comfort Noise Generation
• G.729, an MOS of about 4.0 • G.729A an MOS of about 3.7
62
Digital Speech Interpolation (DSI)
• Voice Activity Detection (VAD)
• Removal of voice silence • Examines voice for power, change of power • Automatically disabled for fax/modem
63
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
64
Voice Quality Comparison
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
3–5 15 15
65