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VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003
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Page 1: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

VOIP 101: The Fundamentals of IP Telephony

William Simmelink, General ManagerVoIP Business UnitTexas Instruments

February 2003

Page 2: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 2

Agenda

Internet Telephony Call Basics

Fundamental Components of VoIP

Gateways

VoIP Applications

Page 3: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 3

Voice Over Internet Protocol (IP)

There are three styles of Voice over IP calls:

Phone to Phone

PC to Phone

PC to PC

InternetIntranet

Gateways adapt traditional telephony to the Internet.

Page 4: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 4

Telephony Signaling

CentralOfficeSwitch

Idle

First Digit is Dialed

DTMF Detector Activated in the CO

Dial-Tone On

Remaining Digits Dialed

Dial Tone Off

Ring Back

Voice-mode

Connected

On-Hook

Off-hook

Signals are exchanged between a telephoneand the switch at the Central Office. Thesesignals connect and disconnect calls as wellas inform the caller of the progress of the call.

Signals are exchanged between a telephoneand the switch at the Central Office. Thesesignals connect and disconnect calls as wellas inform the caller of the progress of the call.

Page 5: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 5

InternetIntranet

Packet Signaling

All three VoIP calls can use H.323, or SGCP/MGCP to set up the Internet portion of the call.

Calls involving gateways must also perform telephony signaling.

Page 6: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 6

Voice over Internet Signaling

Sending voice over a data network requires advanced signaling techniques in the gateways.

InternetIntranet

CentralOfficeSwitch

The gateway connected to the central office must emulate the telephone.

The gateway connected to the phone must emulate the signaling functions of the central office.

Page 7: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 7

Voice over Internet Signaling

Telephone numbers are translated to data network addresses (Internet addresses).

InternetIntranet

CentralOfficeSwitch

Telephony signals are interpreted by the gateway and mapped to the appropriate network protocol (H.323/SGCP/MGCP for IP) set-up, maintenance, billing and tear-down messages.

Page 8: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 8

PBXTelephone

DSP MICRO MICRO DSP

Off-hook

Voice-modevoice mode

On-hook

Idle-mode

Off-hook

DTMF Modefirst digits

Dial-tone

digits

digitsdigits

idle mode

Dial-tone off

Dial-tone

Switched CAS (FXS-FXO)

Network

connectconnect_ack

release

setup

setupcall_proceeding

call_proceeding

connectconnect_ack

Call Progress In Band

release

H.323SGCP/MGCP

Page 9: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 9

Fundamental Components of VoIP Gateways

Page 10: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 10

Micro Ethernet(Internet)

Micro Processor(s) Telephony Protocols Network Protocols Management Routing Billing

How is it all Done?

Within the Gateway a series of processors perform the adaptation from Traditional to Internet Telephony.

DSP

DSP

DSP

DSP

Telephones(Circuits)

Digital Signal Processor(s) (DSP) Voice Compression Tone Detection/Generation Echo Cancellation Silence Suppression

Page 11: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 11

Analog Voice to PCM

An analog voice signal is received.

The Signal is converted to a Pulse Code Modulation(PCM) digital stream.

10110101 11010011 11001001 00100100 00111100 10010011 11100001 00100100 00111100 10010011 10110101 11010011 11001001 00100100 00111100 10010011 11100001 001

DSP

Page 12: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 12

10110101 11010011 11001001 00100100 00111100 10010011 11100001 00100100 00111100 10010011 10110101 11010011 11001001 00100100 00111100 10010011 11100001 00100100

PCM Processing

The PCM stream is analyzed.

DSP

Detected signaling tones are routed around the CODEC. (needed, since most CODECs garble signaling tones to the point that they are unrecognizable)

Tone Detection is performed:

Echo is removed.

The Voice Activity Detector (VAD) removes silence.

Remaining stream is passed to CODEC.

Page 13: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 13

PCM to Frames

11010011 11001001 00100100 00111100 10010011 11100001 00100100 00111100

. . . and voice frames are created

10110101 11010011 11001001 00100100 00111100 10010011 11100001 00100100 00111100

Most CODECs also compress the PCM stream: PCM G.711 generates 64,000 bits per second G.729a compression generates 8,000 bits per second

DSP

The PCM stream is fed into the CODEC . . .

10110101

Each Frame is 10 ms long (G.729a) and contains 10 bytes of “speech.”

Page 14: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 14

Frames to Packets

DSP

10110101

Packet Assembler Software within the DSP takes frames from the CODEC and creates packets.

The packet is forwarded to the gateway’s host processor.

Several frames may be combined in a single packet

10110101 10110101 10110101RTP

A 12 byte Real Time Protocol (RTP) Header is added: Provides sequence number Time stamp

Page 15: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 15

IP

A 20 byte IP header is added to the packet containing: The IP address of this gateway (the source address) The IP address of the destination gateway

An 8 byte UDP header containing source and destinationsockets is also added.

UDP

Addressing

Dialed digits identified by the tone detection performed in the DSP are used to determine the destination number.

1011010110110101 1011010110110101RTP

301-999-1212

This number is mapped to an IP Address.

= 192.128.100.2

Micro

Page 16: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 16

In the Internet

Routers and Switches in the Internet examine the addressesin the IP address in order to identify the route to the destination.

Several routers and or switches may be in the path thatthe packets take to their destination.

Page 17: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 17

IP

Upon Arrival at the Destination

The IP and UDP headers are removed from the packet in the Microprocessor.

UDP

Micro

RTP

The Packet is forwarded to the DSP where theRTP Header is removed.

Finally, the packet is disassembled leaving thevoice frames.

1011010110110101 10110101 10110101

Page 18: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 18

Various Network Problems are Dealt With

Voice Packets are generated at a constant rate while someoneis speaking; there is essentially no gap between packets.

These gaps, known as jitter, must be removed by the receivinggateway in order to accurately reproduce the original speech

Devices in the network cause an unpredictable amount ofdelay to occur between packets.

Page 19: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 19

Jitter Removal

An adaptive jitter buffer in the receiving DSP is used to smooth the playout of packets arriving from a “jittery” network.

DSP

DSP

This eliminates the jitter induced distortion that would have been heard by the listener.

Page 20: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 20

Lost Packets

Congestion in the network may cause some packets to be dropped.

1

2

4

6

5

3

Left untreated, the listener hears annoying pops & clicks.

Page 21: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 21

Lost Packets

An algorithm in the DSP detects missing packets.

1

2

4

6

And replays the last successfully received packet at a decreased volume in order to fill the gaps.

1

2

4

6

4

2

DSP

35

Page 22: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 22

Turning “Hello”…..

Into “oHell”

Out of Order Packets

Out of order packets are not played in the order they arrive…..

Packets may take diverse routes through a network and may arrive out of order.

1

2

4

5

3

DSP

Page 23: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 23

Out of Order Packets

1

2

4

5

DSP

When an out of order condition is detected the missing packet is replaced by its predecessor as if it is lost.

2

When the late packet finally arrives it is discarded.

3

Page 24: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 24

PCM Back to Analog

A Comfort Noise Generator fills in the gaps that were created during silence detection and suppression.

The PCM Stream is reconstituted as an analog signal and is played out to the listener.

10110101 11010011 11001001 00100100 00111100 10010011 11100001 00100100 00111100 10010011 10110101 11010011 11001001 00100100 00111100 10010011 11100001 00100100

Page 25: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 25

VoIP Applications

Page 26: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 26

Central Office/Infrastructure

CentralOffice Gateway

PacketNetwork

Traditional carriers migrate to packet core for lower network costs.

Gradual capping of Class 4 tandem switches drives CO/Infrastructure VoIP ports.

Carriers proposing new packet architectures with dramatically lower cost structures.

Page 27: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 27

Enterprise

SME Gateway

IP Phone

PBX

PacketNetwork

Enterprises deploying to avoid access charges and settlement fees.

Businesses take advantage of existing data networks.

Reduced operating costs by managing one network.

Page 28: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 28

IP Phones and PBX Trunking

Office 2Office 2

Office 1Office 1

T1

PacketNetwork

Gateway

Router

IP Phone

PBX

LAN-based PBX for cost reduction, flexibility, and new applications: Integrated voice/data LAN infrastructure Integrated voice/data applications Open hardware platform

Page 29: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 29

Residential Broadband

Residential voice alternatives, leveraging broadband connections

VoCable solutions in trials in US, and deployments in Europe

VoDSL deployments in Asia and Europe

Fiber to the Home potential in China

CMTS DSLAM

CPE Gateway Cable

Modem

Packet Network

Cable or DSL Modem Based IAD

VoiceGateway

Page 30: VOIP 101: The Fundamentals of IP Telephony William Simmelink, General Manager VoIP Business Unit Texas Instruments February 2003.

Page 30

Summary

VoIP solutions require well integrated, robust set of functional components for toll quality operation.

VoIP implementations are in current systems deployed worldwide.

VoIP value proposition exists in different vertical markets.


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