Voice over IP Protocols And Compression Algorithms

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Voice over IP

Protocols And Compression Algorithms

Presented by:

Neda Kazemian Amiri

University of Tehran

Electrical and Computer Engineering School

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Agenda

Introduction to VoIP

Protocols

Compression Algorithms and

Standards

z

G. Standards

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• Voice existed on proprietary,

circuit-switched

telephone networks

Like two parallel universes, data and voice were

in the same space on different planes

Like two parallel universes, data and voice were

in the same space on different planes

(4)

Cost initially...but emerging business

applications will be the real driver

z

Integrated e-mail/voice

mail messaging

z

Web-based call centers

z

Computer Telephony

Integration (CTI)

z

Intranet and Internet

telephony

z

Fax

Data/Voice

(5)

One Infrastructure

Data, Voice, and Video

Integration Benefits

Provides data, voice and video

“Voice rides for free”

PBX elimination

Simplified operations

Budget leverage

(6)

• Personal Rules Engine

–

Administration/

filtering

–

Visual messaging

–

Calendar setting

–

Integrated search and

Cell/PDA

(7)

Traditional TDM Networking

TDM uses dedicated bandwidth to establish an

end-to-end connection

Traffic entering a TDM device must be less than or

equal to the amount of traffic leaving the TDM device

(8)

(9)

Statistical multiplexing makes efficient

use of bandwidth

Input traffic into the statistical device

may be greater than the output traffic

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Network

1. Speech is converted

to digital voice

packets

2. Packets are sent

over a data network

3. Digital voice packets

are converted to

analog speech

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1. Person speaks

into telephone

CODEC

2. CODEC (coder-decoder)

converts the signal from analog

to digital data packets suitable

for transmission over a TCP/IP

data network

3. Digital signal processing

(DSP) chip compresses the

packets for transmission

over the data network

DSP

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Converting from Data

Back to Voice

6. Recipient listens to the

voice on their telephone

5. CODEC software converts the

signal from digital data packets

back to analog voice

4. DSP chip uncompresses

the packets

CODEC

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Choosing VoIP

Converged Networks

z

Voice, Video & Data over an IP network

z

Reduced the costs of managing parallel networks

z

Allows voice to be an IP “application”

z

Handles phone-to-computer voice communications

z

unified messaging

z

Web-enabled call centers

Centralized or distributed architectures

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Basic Components of VoIP

Voice Processing

z

Codec (Coder/Decoder)

z

Echo Cancellation unit

z

Voice Activity/Idle noise detector

Call Processing

z

PSTN interface

z

Signaling protocol management

z

Call establishment/teardown

z

Telephone number mapping

Packet Processing

z

Convert data stream from codec to packet format

z

Adds appropriate transport headers

z

Convert signaling protocols: Telephony to packet signaling

Network Management

z

Performance monitoring

z

Fault detection

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Silence Suppression by

Voice Activity Detection

Speech

Silence

Speech

No Cells

Cells

Voice Activity Detection (VAD)

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(17)

Definition of Protocol

A protocol is a set of formal rules

which describe how to transmit

data.

Protocols may deal with the way

data is formatted, including:

z

syntax of messages

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VoIP Unions Providing

Standard

z

ITU-T:

International Telecommunications Union

z

International standards body for telephony

z

ITU-T H.323—International Telecommunications Union

recommendation for multimedia (including voice) networking

over IP

z

IMTC:

International Multimedia Teleconferencing

Consortium

z

International standards body providing recommendations for

multimedia networking over IP, including VoIP

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Voice over IP

Components

H.323 standard

z

Specifies call setup and interoperability

z

G. standards

z

Specify analog-to-digital conversion and compression

Realtime Transport Protocol (RTP)

z

Manages end-to-end connections to minimize the

effect of packets lost or delayed in transit

Internet Protocol (IP)

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Protocols

H.323 z

_{an ITU standard}

z

describes how multimedia communications occur

between terminals, network equipment, and

assorted services on IP networks.

SIP (Session Initiation Protocol)

z

a competing IETF standard

z

performs similar functions to H.323

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H.323 – A Closer Look

Terminals (what people see/hear)

Gateways (control and ‘routing’ )

Multipoint Control Units (provides

conference capabilities )

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Terminals

H.323 client endpoints

They could be:

z

Multimedia PCs

z

Any stand-alone device

z

A simple telephone

Expectation by H.323:

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Gateways

Optional Component of H.323

implementation.

Used as interface between different networks

e.g. LAN & PSTN

Functions:

z

Data format translation

z

Audio/video codec translation (DSP’s)

z

jitter buffers, echo cancellation, and packet

processing

z

Call setup, termination from both sides of

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Multipoint Control Units

MCUs are also optional

in a H.323

implementation

Needed only when

multiparty conferences

are desired

Functions:

z

Provides capability of

video-conferencing with

more than one party.

z

_{Acts as a coordinator of}

(25)

Gatekeepers

“Brains” of a H.323 network

(26)

The Protocols

H.225.0 defines the call

signaling between endpoints

and the Gatekeeper

RTP/RTCP

is used to

transmit media such as audio

and video over IP networks

H.225.0 define the

procedures and protocol for

communication within and

between Peer Elements

H.245 is the protocol used to

control establishment and

closure of media channels

within the context of a call and

H.450.x

is a series of

supplementary service protocols

H.460.x

is a series of

version-independent extensions

to the base H.323 protocol

T.120 specifies how to do

data conferencing

T.38 defines how to relay fax

signals

V.150.1 defines how to relay

modem signals

H.235 defines security within

H.323 systems

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Characteristics of H.323

Centralized and Distributed Control

z

Endpoints can communicate directly or through a server

Focused

z

_{provides users with voice, data, and video conferencing over}

a packet-switched network

Extensibility

z

Fully backward compatible with previous versions.

z

Equipment manufactures can extend functionality by

inserting their own additions to the protocol.

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5

4

3

2

1 Su

bjective Quality (MOS)

Hybrid Coders

(LD-CELP and

CS-ACELP)

Vocoders

(Older Technology)

Waveform Coders

(ADPCM)

Score

Score QualityQuality Description of ImpairmentDescription of Impairment 5 5 4 4 3 3 2 2 1 1 Excellent Excellent Good Good Fair Fair Poor Poor Bad Bad Imperceptible Imperceptible

Just Perceptible, Not Annoying

Perceptible and Slightly Annoying

Annoying but Not Objectionable

Very Annoying and Objectionable

Voice Quality Guidelines

(29)

Linear Predictive Coding

and Speech Synthesis

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a) Coder and b) Decoder [3]

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Audio:

z

PCM-Based: G.711, G.721, G.722, G.726

z

CELP-Based: G.723.1, G.728, G.729

Video

z

H.261 codec (for channels with bandwidths

p*64 kb/s)

z

H.263 codec (for low bit rate transmission

without loss of quality)

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ITU G. Standards for Audio

G.711

z The first standard for speech compression. z PCM, Frame size = 0.125ms z 64Kbps z used in PSTN z MOS=4.1

G.721

z ADPCM, Frame size = 0.125ms z 32Kbps

z used in PSTN.

z Year of introduction: 84

G.722

z It is like G.721.

z maximum bit rate 64Kbps.

G.726

z ADPCM, Frame size = 0.125ms z 16, 24, 32 and 40 Kbps

z MOS=3.85

G.723.1

z hybrid coder, Frame size = 30ms

z with MP-MLQ algorithm its bit rate is 6.3 Kbps. (MOS=3.9)

z with ACELP algorithm its bit rate is 5.3 Kbps. (MOS=3.65)

z used for videophones.

G.728

z hybrid coder

z LD-CELP algorithm, Frame size =.625ms

z 16 Kbps.

z uses 5 samples frames. z MOS=3.61

G.729

z hybrid coder

z CS-ACELP algorithm, Frame size = 10ms

z 8 Kbps

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LD-CELP Decoder Used

in G.728

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Future Work

Studying & comparing the VoIP Protocols in

more detail

z

H.323 z

SIP

z

MGCP

(Media Gateway Control Protocol)

z

Skinny

( a cisco standard)

Studying CELP-based compression algorithms

and codec recommendations

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References

[1] Cisco systems,

http://www.cisco.com

[2] Swale, R., “Voice over IP: systems and solutions,”

The Institution of Electrical Engineers, 2001

[3] ITU-T Recommendation G.729, “Coding of Speech

at 8Kbps Using Conjugate-Structure Algebraic Code

Excited Linear Prediction (CS-ACELP),” 1995

[4] ITU-T Recommendation G.723.1, “Dual Rate

Speech Coder for Multimedia Communications

Transmitting at 5.3 and 6.3Kbps,” 1996

[5 ITU-T Recommendation G.728, “coding of Speech at

16Kbps Using Low-Delay Code Excited Linear

Prediction,” 1992

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[7] Xydeas, C., “An Overview of Speech Coding Techniques”,

Speech coding – Techniques and applications, IEE colloquium

on 14 Apr. 1992, pp. 111 – 125.

[8] Kipper, U., Reininger, H., and Wolf, D., “CELP Coding with

Adaptive Excitation Codebooks”, IEEE., 1991.

[9] Schroeder, M. R., and Atal, B. S., “Code-Excited Linear

Prediction (CELP): High-Quality Speech at Very Low Bit Rates”,

IEEE, 1985.

[10] Owen, F.E., “PCM and Digital Transmission Systems”,

McGRAW-HILL Book Company, 1976.

[11] Proakis, J. G., Salehi, M., “ Contemporary Communication

Systems Using MATLAB”, PWS Publishing Company, 1997

[12] Brunner, S., and Ali, A. A., “Voice over IP, Understanding VoIP

Networks”, Juniper Networks, Inc., 2004. www.juniper.net

[13] M. Banerjee, B. A. Vani, S. Madhusudhan and S. Monga,

”Optimizations of ITU G.729 Speech Codec,” IEEE 60th

Vehicular Technology Conference, vol. 6, pp. 3913 – 3918,

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