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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!
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