Basic VOIP

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Basic VOIP
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




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