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Optimizing Converged Cisco Networks (Ont) : Module 2: Cisco Voip Implementations

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25 views

Optimizing Converged Cisco Networks (Ont) : Module 2: Cisco Voip Implementations

Uploaded by

ccazorlaqsc
Copyright
© © All Rights Reserved
Available Formats
Download as PPT, PDF, TXT or read online on Scribd
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Optimizing Converged

Cisco Networks (ONT)

Module 2: Cisco VoIP Implementations

© 2006 Cisco Systems, Inc. All rights reserved.


Lesson 2.2:
Digitizing and
Packetizing Voice

© 2006 Cisco Systems, Inc. All rights reserved.


Objectives
 Describe the process of analog to digital conversion.
 Describe the process of digital to analog conversion.
 Explain how sampling rates are determined using the
Nyquist Theorem.
 Explain how quantization can lead to noise.
 Explain how MOS is used to judge voice quality.
 Describe the purpose of DSPs.

© 2006 Cisco Systems, Inc. All rights reserved.


Basic Voice Encoding: Converting Analog
Signals to Digital Signals

 Step 1: Sample the analog signal.


 Step 2: Quantize sample into a binary expression.
 Step 3: Compress the samples to reduce bandwidth.

© 2006 Cisco Systems, Inc. All rights reserved.


Basic Voice Encoding:
Converting Digital Signals to Analog Signals

 Step 1: Decompress the samples.


 Step 2: Decode the samples into voltage amplitudes, rebuilding
the PAM signal.
 Step 3: Reconstruct the analog signal from the PAM signals.

© 2006 Cisco Systems, Inc. All rights reserved.


Determining Sampling Rate with the Nyquist
Theorem
 The sampling rate affects the quality of the digitized signal.
 Applying the Nyquist theorem determines the minimum sampling
rate of analog signals.
 Nyquist theorem requires that the sampling rate has to be at least
twice the maximum frequency.

© 2006 Cisco Systems, Inc. All rights reserved.


Example: Setting the Correct Voice Sampling
Rate
 Human speech uses 200–9000 Hz.
 Human ear can sense 20–20,000 Hz.
 Traditional telephony systems were designed for
300–3400 Hz.
 Sampling rate for digitizing voice was set to 8000
samples per second, allowing frequencies up to 4000
Hz.

© 2006 Cisco Systems, Inc. All rights reserved.


Quantization
 Quantization is the representation of amplitudes by a
certain value (step).
 A scale with 256 steps is used for quantization.
 Samples are rounded up or down to the closer step.
 Rounding introduces inexactness (quantization noise).

© 2006 Cisco Systems, Inc. All rights reserved.


Quantization Techniques
 Linear quantization:
Lower SNR on small signals (worse voice quality)
Higher SNR on large signals (better voice quality)

 Logarithmic quantization provides uniform SNR for all


signals:
Provides higher granularity for lower signals
Corresponds to the logarithmic behavior of the human ear

© 2006 Cisco Systems, Inc. All rights reserved.


Digital Voice Encoding
 Each sample is encoded using eight bits:
One polarity bit
Three segment bits
Four step bits

 Required bandwidth for one call is 64 kbps


(8000 samples per second, 8 bits each).
 Circuit-based telephony networks use TDM to combine
multiple 64-kbps channels (DS-0) to a single physical
line.

© 2006 Cisco Systems, Inc. All rights reserved.


Companding
 Companding — compressing and expanding
 There are two methods of companding:
Mu-law, used in Canada, U.S., and Japan
A-law, used in other countries

 Both methods use a quasi-logarithmic scale:


Logarithmic segment sizes
Linear step sizes (within a segment)

 Both methods have eight positive and eight negative


segments, with 16 steps per segment.
 An international connection needs to use A-law; mu-to-
A conversion is the responsibility of the mu-law country.

© 2006 Cisco Systems, Inc. All rights reserved.


Coding
 Pulse Code Modulation (PCM)
Digital representation of analog signal
Signal is sampled regularly at uniform levels
Basic PCM samples voice 8000 times per second
Basis for the entire telephone system digital hierarchy

 Adaptive Differential Pulse Code Modulation


Replaces PCM
Transmits only the difference between one sample and the next

© 2006 Cisco Systems, Inc. All rights reserved.


Common Voice Codec Characteristics
ITU-T
Codec Bit Rate (kbps)
Standard

G.711 PCM 64

G.726 ADPCM 16, 24, 32

G.728 LDCELP (Low Delay CELP) 16

G.729 CS-ACELP 8

CS-ACELP, but with less


G.729A 8
computation
© 2006 Cisco Systems, Inc. All rights reserved.
Mean Opinion Score

© 2006 Cisco Systems, Inc. All rights reserved.


A Closer Look at a DSP
A DSP is a specialized processor
used for telephony applications: DSP Module

 Voice termination:
Works as a compander converting
analog voice to digital format and
back again Voice Network Module
Provides echo cancellation, VAD,
CNG, jitter removal, and other
benefits

 Conferencing: Mixes incoming


streams from multiple parties
 Transcoding: Translates between
voice streams that use different,
incompatible codecs
© 2006 Cisco Systems, Inc. All rights reserved.
DSP Used for Conferencing
 DSPs can be used in
single- or mixed-mode
conferences:
Mixed mode supports
different codecs.
Single mode demands that
the same codec to be used
by all participants.

 Mixed mode has fewer


conferences per DSP.

© 2006 Cisco Systems, Inc. All rights reserved.


Example: DSP Used for Transcoding

© 2006 Cisco Systems, Inc. All rights reserved.


Self Check
1. What sampling frequency is recommended by the
Nyquist Theorem for reconstruction of a signal?
2. What is the Hz range for traditional telephone
systems?
3. What is the implication of using 8 bits for
quantization?
4. What is the purpose of logarithmic quantization?
5. What is MOS?

© 2006 Cisco Systems, Inc. All rights reserved.


Summary
 Voice-enabled routers convert analog voice signals to
digital format for encapsulation in IP packets and
transport over IP networks. These packets are
converted back to analog at the other end.
 Quantization is the process of selecting binary values to
represent voltage levels of voice samples. Quantization
errors arise when too few samples are taken.
 There are two methods of companding: Mu-law, used in
Canada, U.S., and Japan, and A-law, used in other
countries.
 The Mean Opinion Score (MOS) provides a numerical
indication of the perceived quality of received media
after compression and/or transmission.
© 2006 Cisco Systems, Inc. All rights reserved.
Q and A

© 2006 Cisco Systems, Inc. All rights reserved.


Resources
 Voice Codec Bandwidth Calculator (requires CCO
login)
http://tools.cisco.com/Support/VBC/do/CodecCalc1.do

 DSP Calculator
http://www.cisco.com/cgi-bin/Support/DSP/dsp-calc.pl

 Free VoIP Quality Tester


http://www.testyourvoip.com/

© 2006 Cisco Systems, Inc. All rights reserved.


© 2006 Cisco Systems, Inc. All rights reserved.

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