Telecommunication

Networks and SystemsNetworks and Systems

Voice service

Krzysztof Wajda

Deparment of Telecommunications, AGH-USTNovember, 2016

Outline

• Integration of voice services

• Voice integration technologies

• Voice coding and compression

• Voice in IP• Voice in IP

• Voice in ATM

• Efficiency of voice transmission

• Conclusions

ATMATM

Background...

• 80-85% incomes in telecom market is

related to classical telephony and basic

services,

• Number os subscribers lines in the world• Number os subscribers lines in the world

exceeded 1 bln (but is decresing due to

proliferation of mobile systems)...

• ... but it is estimated that also approx. 1 bln

people never used telephone set

Why integration of voice service?

PBX PBX

telephone

network\

RouterRouter

data

network

Outcomes from integration

Lowering exploitation costs (OPEX).

Simplified administration and

management.management.

Hiher efficiency of resource usage.

Flexibility of solutions.

Technologies fof voice integration

IP

ATMATM

Frame Relay

Basic voice signal encoding - PCM

Voice sampling - 8 kHz,

Each sample is encode using 8 bits, Each sample is encode using 8 bits,

Basic PCM channel 64 kbps.

Voice compression, silence removal

Transmitted

signal

Received

signalvoice

noise

Voice activity

detector VADNoice generator

CNG

synchronization

of modules

When using voice compression and

silence removal we use also noicve

genarator at the remote end (for

listener convenience).

Voice compression methods

AlgorithmVoice quality

(P.800)

Bandwidth

(kbps)MIPS (nakład

)

Całkowite opóźnienie

(ms)Aplikacja

PCM 4.11 64 - 0.25 PSTN

ADPCM

(G.726)3.85 32 10 0.25

PSTN, Mobile telephony(G.726)

CS-ACELP

(G.729)3.92 8 30 25

Głos na FR,

ATM, IP

CS-ACELP

(G.729A)<3.92 8 20 25

Voice in FR,

ATM, IP

LD-CELP

(G.728)3.61 16 40 1.25 PSTN

MP-MLQ

(G.723.1)- 5.3/6.3 30 67.5 Voice in IP

Voice compression quality and its correlation with bandwidth

64P

rzepły

wność (

kbps)

nieakceptowalna biznesowa wysoka

PCM

Jakość głosu0

8

16

24

32

Prz

epły

wność

ADPCM 32 (G.723)

ADPCM 24 (G.725)

LDCELP 16 (G.728)ADPCM 16 (G.726)

LPC 4.8

CS-ACELP (G.729a)

CS-ACELP (G.729)

za Cisco Systems

Echo

sources:

Chnging transmission lines from 4 to

2 wires,

Crosstalk in analog transmissionCrosstalk in analog transmission

lines,

not matching elements (impedance),

Acoustic echo in rooms.

Necessity to eliminate echo for delays

greater than 25 ms.

Quality requirements for voice transmission

• mean delay,

• delay jitter,• delay jitter,

• losses.

Reasons of delay

• Encoding delay,

• waiting time for filling of transmission frame,

• delays from data computation (routing,

switching, buffering),

• delays in internetworking module (IWF,

gateway),

• propagation delay.

Voice service in IP

PROs

• ubiquity of IP,

• low cost of

CONs

• low quality of voice,

• many standards,

connections,

• integration with

WWW,

• independence from

physical medium,

•standard adressing,

• possible overload –

with loss of

information.

Hardware requirements

•Typical processor,

• audio card,

•microphon and headphones,

• network card.

Connection types

• Direct among terminals,

• with signalling server,

• with proxy server,

• with server and PBX.

Voice in ATM

Scalability

QoSQoS

•CBR

•rt-VBR

Types of settings in ATM

Desktop

PBX

•structured•structured

•unstructured

POTS in ATM

• not popular (lack of dedicated,

•ATM is not suitable for low speed services (like

voice) due to delays for packetization and

assemblyingassemblying

• in the future universal multimedia terminals

based on IP.

Network integration

UNIPrywatna

Private

N-ISDNIWF

PNNI or UNI

Q.SIG

UNI

UNIPrywatnasieć ATM Public

N-ISDN/PSTN

PublicATM network

UNI

IWF

PublicUNI

UNIPNNIIWFQ.SIG

PRIBRI

BRI or PRI

User Network InterfacePrivate Network Network InterfaceInterworking FunctionN-ISDN signalling

Primary Rate InterfaceBasic Rate Interface

Voice transmission with AAL1

ATM ATM ATM ATM

audio stream

N N N N

1 5 47 B 1 5 47 B 1 5 47 B 1 5 47 B

Streaming type of transmission

AAL1 introduces small overhead (6/53)

Built-in synchronization in AAL1

Packetization delay 6.5 ms

ATMATM

8 B40 B of samples

NN

5 48 B5 48 B

8 B40 B of samples

Voice transmission with AAL5

Voice transmission using AAL5 is efficient

(not significant overhead)

assures detection and correction of errors

using AAL5

Packetization delay 5.0 ms

5 48 B

channel

1

voice

channel 16data

channel

1

voice

channel 163AAL2 User Part

AAL2 Common

Part

Voice and data transmission with AAL2

ATM cell ATM cellATM

Cell header Cell header

Multiplexing in single VC

One-stage multiplexing of voice channels

Parameters of single voice source

ON phase (caller is active) – exponential

distribution with average value 350 ms,

OFF phase (“silence”) - exponential

distribution with average value 650 ms,distribution with average value 650 ms,

in ON phase a source generates 1 B each

125 ms (PCM),

in OFF a source is silent.

Quality parameters

Max. delay: 150 ms

Acceptible loss level: Acceptible loss level:

1%

0.1%

Estimated number of telephone channels (1)

86

65

84

63

50

60

70

80

90

No of channels

256384

5121024

15362048 p=0,1% for t=150 ms

p=1% for t=150 ms

38

97

4

36

86

4

0

10

20

30

40No of channels

Link throughput [kbps]

Delay[ms]Bandwidth

[kbps]

Numberof TDM (PCM channels g

channels

Loss 1% Loss 0,1 %

256 4 4 168 200

Estimated numbr of telephone channels (2)

Number

of ATM voice

256 4 4 168 200384 6 7 81 113384 6 8 112 191512 8 11 89 1301024 16 25 45 711024 16 26 64 1642048 32 (30) 61 129 1482048 32 (30) 62 149 242

Delay vs multiplexing gain

400

500

600

ms]

1024 kbps - 0,1%

1024kbps - 1%

2048kbps - 0,1%

2048kbps - 1%

Próg 150ms

0

100

200

300

200% 220% 240% 260% 280% 300% 320%

Multiplexing gain

De

lay

[ms

Results show importance

of traffic consolidation in

PBX installations

• ASX200 – configuration manual•

Bibliography

Thank you!