Emerging Technology Development for Multimedia Communications

John P. YuNovember 13, 2006

Slide 1 of 48 Confidential

Emerging Technology Development for Multimedia Communications

John Pang Yu Ph. D.

Managing DirectorThe Emerging Technologies Cooperative, Inc.

Menlo Park, California USA

National Taipei University of Technology, Taipei, Taiwan



Table of Contents (Continued)

• DCoIP Device and System Developers

• Existing and Proposed Standards

• Quality of Services

• Technical Advancements

• DCoIP Market Trends

• DCoIP Challenges

• References



Introduction To Conventional Communication Technological Concepts

• Transmission Line Loss, Echoes, Delay, Gain

• Insertion Loss, Impedance Matching, Crosstalk, Return Loss

• Signal Bandwidth and Noise Measurement

• Differential Gain and Differential Phase (non-linearity of a two-port system)

• X.25 Seven Layer Protocol: Application, Presentation, Session, Transport, Network, Data Link and Physical

• Modulation:

Amplitude-Modulated Signals: Double-Sideband with Transmitter Carrier; Double-Sideband with Suppressed Carrier; Single-Sideband with Transmitter

Carrier.



Introduction To Conventional Communication Technological Concepts (Continued)

• Modulation:

Amplitude-Modulated Signals: Vestigial-Sideband with Transmitter Carrier (TV RF Signal)

Angle-Modulated Signals; Pulse Amplitude Modulation; Pulse Duration Modulation; Pulse Code Modulation; Frequency-Modulated Signals; Digital Subscriber Line (DSL)

• Multiplexing:

Space Division Multiplex (SDM); Frequency Division Multiplex; Time-Domain Multiplex (TDM); Wavelength Division Multiplex (WDM); Digital Subscriber Loop (DSL) Access Multiplex (DSLAMs)



Some Data Communication Interface Standards

• Committee Consulting International Telephone and Telegraph (CCITT)

• Electronic Industrial Association (EIA): RS-232-C, RS-422,

RS-485

• PSTN, ISDN, DSL, DSLAM, T1 (DS-1: 1.544 Mbps), E1 (2.048 Mbps), OC-1 (STS-1: 51.84 Mbps)

• T.38: Voice, Data and Fax over Different Telephone Lines

• Loop Start Interface (LSI: analog PSTN)



Interactive Video Display System

• A Bi-Directional Interactive Data Communication Systems Via MODEM and Telephone Line

• Image Plans and video Graphic Mode

• Texts and Graphics Mixed Mode

• Video Graphics and Texts Display Processors in A Digital Format

• Information Retrieval Between Video Display Terminal and Terminal

• Information Retrieval Between Video Display Terminal and Database (Information Provider)

• A Bi-Directional Interactive Data Communication System Via RF MODEM and Cable



Worldwide Video Standards

NTSC PAL SECAM

Line / Field 525 / 60 625 / 50 819 / 50 “E” Mono

625 / 50 “L” Color

H. Frequency 15.734 KHz 15.625 KHz 20.745 KHz “E”

15.625 KHZ “L”

V. Frequency 59.94 Hz 50 Hz 50 Hz “E” & “L”

Color Subcarrier 3.579545 MHz 4.433618 MHz 4.40625 MHz OR

4.25000 MHz OB

Sound Carrier 4.5 MHz (FM) 6.0 MHz (FM) 6.5 MHz (AM) “L”

Video Bandwidth (Y) 4.2 MHZ 5.5 MHz 10 MHz “E”

6.0 MHz “L”

Video Component R G B Or R G B Or R G B Or

Y I Q or Y U V Y U V

Y B-Y R-Y

Interlaced 2 : 1 2 : 1 2 : 1

Frames / Second 30 25 25

Aspect Ratio 4 : 3 4 : 3 4 : 3



Proposed HDTV Standards

Japan USA Europe

Line / Field 1125 / 60 1050 / 59.94 1152 / 50

H. Frequency 33.7495 KHz 31.468 KHz 31.25 KHzV. Frequency 60 Hz 59.94 Hz 50 Hz

Video Bandwidth (Y) 30 MHz 40 MHz

Chrominance BW (B-Y) 15 MHz 20 MHzChrominance BW (R-Y) 15 MHz 20 MHz

Interlaced 2 : 1 2 : 1 2 : 1Frames / Second 30 30 25 Aspect Ratio 16 : 9 16 : 9 16 : 9



Telecommunications Digital Transmission Hierarch Highlights Digital Optical Electrical Line Effective # DS0s in #DS1s in #DS3s Others SDH

Signal Transmit Transmit Bit Rate Data Rate Payload Payload in Payload Level

DS-0 E0 /J0 64 Kbps 64 Kbps 1

DS-1 T1 /J1 1.544 Mbps 1.536 Mbps 24 1

E1 2.048 32

DS-2 T2 6.312 96 4

E2 8.448 128

E3 34.368 512

DS-3 T3 44.736 672 28 1

OC-1 STS-1 51.84 50 672 28 1

E4 139.264 2048

OC-3 STS-3 155.52 150 2016 84 3 STM-1

DS-4 274.176 4032 168 6

OC-9 STS-9 466.56 451 6048 252 9 STM-3

OC-12 STS-12 622.08 601 8064 336 12 4 OC-3 STM-4

OC-24 STS-24 1.244 Gbps 1.20 Gbps 16128 672 24 STM-8

OC-96 STS-96 4.976 4.81 64512 2688 96 STM-32

OC-256 13.271 172032 7168 256

OC-768 39.813 516096



Introduction to Digital Contents over IP

The Adoption and Implementation of Digital Content over IP Can be Justified by the Following:

Increasing Voice/Data/Video Convergence IP is Now the “Common Protocol”; RSVP Protocol for

Bandwidth Reservation and RTP Protocol for Detecting Missing Packets to Improve Quality of Services

Packetized Compressed Voice Has Shown Cost-Effective Solutions (High-Class Coding Algorithms)

Intranets and Extranets are Growing Rapidly Voice Over Frame Relay is Being Successfully Deployed in

Major Corporate Network The Rapid Growth of Digital Multimedia Contents in Internet



DCoIP Network Topology

Equipment to Bridge the Circuit-Switched Networkand Packet-Switched Network

Gatekeeper

IPNetwork

PSTN PSTN

PSTN to VoIPVoIP to PSTN

GatewayPSTN to VoIPVoIP to PSTN

Gateway

IP NetworkConnection

PhoneLine

PhoneLine

PhoneLine

PhoneLine






Router Router

H.323 MCS with

gateway

H.323 MCS with

gatewayH.323 Gatekeeper

Firewall & H.323 proxy

H.323 EndPoints

H.323 EndPoints

H.323 EndPoints

H.323 EndPoint

H.323 EndPoint

Telephone

INTERNET

ISDN POTS

Circuit Switched Network




Major Entities in an H.32X Environment: H.323 Terminals, Gateways, Gatekeepers and MCUs.

H.320 ISDN

H.324 PSTN

GateKeeperMCU H.323

H.323

H.323

H.323H.323

Gateway

Circuit Switched Network Internet

H.323Terminals

H.323Terminals

ITUTerminals

H.323 Zone



Circuit-Switched Network

Characteristics - Constant Bit Rate; Full Bandwidth After Call Setup; Low Latency, Constant Delay; Incremental Bandwidth Available (Add B Channels)

Protocols - ISDN, Robbed Bit Signaling Medium - T1 / Fractional T1, DDS at 56 kbps Lines Addressing Schemes - Use Phone Numbers; Statically

Assigned; Public Directory Assistance if Unknown H.320 Terminals Intended for Voice Transmission Data Transmission Using Modems High Quality connections – Low Delay, High

Reliability, full Duplex 56 / 64 Kbps Channels Basis for Toll Quality



Packet-Switched Network

Characteristics - Burst Mode; Variable Bit Rate; Variable Latency and Variable Delay; Non-Guaranteed Quality of Services in Current Network Topologies; Incremental Bandwidth Quickly Becoming More Available

Protocols - TCP/IP, IPCP (Internet Protocol Control Protocol), DHCP (Dynamic Host Configuration Protocol)

Medium - Ethernet, Frame Relay, ATM Networks and TCP/IP Transport

H.323 Terminals Addressing Schemes - IP Address; Static or Dynamic Assignment;

Directory Servers - Maintain User IP Address by Name or Alias; IP Address Can Change Depending on Your Location



Internet Protocol Version 6 (IPv6)

New Version of Internet Protocol Used 128-Bit Address and Allow Embedding IEEE 802

Address Flow Label to Identify Real Time and Special Handling

Traffic Increased Scalability for Network Architectures Improved Security and Data Integrity Autoconfiguration Multicast Support Standardized



IPv4 vs IPv6

• Address space increased from 32 bits to 128– IPv4 has about 4 Billion addresses (US, Europe, rest of the world)– IPv6 has about 2^128 = 3.4 X 10 ^ 38 addresses

• Approx. 665 X 10 ^21 addresses per sq.m of the earth surface

• IPv6 has built in IP security (IPsecurity is part of IPv6)• IPv6 has fixed length header.

– Optimized for hardware implementation

• IPv6 has improved support for QoS, Multicast and Mobile IP• IPv6 has support for domestic appliances• Government (DOD 2008), University and Industry lead

initiatives



IPv6 Partners

http://www.yottayotta.com/

http://www.sun.com/



IPv6 Partners (Continued)



IPv6 Partners (Continued)



Major System Components

Terminals Bi-Directional Real-Time Communication for the User Enable Voice Connections, Video and Data

Communication are Optional Supporting H.245 for Describing the Negotiation of the

Appropriate or Required Terminal Functionality

Gateways Translate Between Physical Media, Network Protocols,

Conferencing Protocols and Addressing Translate Between Different Audio or Video Codes in

Real Time, Allow H.320 Endpoints to Use T.120Conferencing Effectively

Support Voice Over IP and Multimedia Conferencing



Major System Components (Continued)

Locate at Multimedia Conference Servers, Stand-Alone Equipment, Network Servers, Routers, Remote Access Servers, Multimedia PBX, Network Services Authorization and Authentication

Serve Conferences Interoperability Between H.320 and H.323 Endpoints Complying With Different ITU Standards, Diverse Network Transport, Diverse Audio or Video Codecs

Connect Incompatible Devices by Device Emulation at the Network Level; a Good Gateway is Invisible Just Like the Gateway Embedded in the Telephone Network

A media gateway provides translation of protocols for call setup and release, conversion of media formats between different networks:

Transfer of information between H.323 and SIP networks on an IP Network

Translation between transmission formats and communication signals and procedures (e.g. between IP and PSTN)

Passes call signaling not applicable to the media gateway through to the network endpoint (e.g. supplemental services such as call forwarding)

Performs call setup and clearing on both sides Translates between encoding formats




Gatekeepers An Optional Element of H.323 Link Endpoints Directly to Gatekeepers Reside Anywhere in H.323 Network Entities or Run as

Server Application Do Bandwidth and Resource Management, Access

Control, Endpoint Registration, Zone Definition,Enhanced Call Control and Address Translation

Platform Independent An Embedded Component in Hardware Building Block Gatekeeper "Engine" Software Development Application




Multipoint Control Units

Provide Audio Bridge With Value-Added VideoMultipoint Capabilities, Unattended Operation, FullAudio Mixing With Advanced Techniques for HighQuality Compressed Speech

Support Multicast and Unicast Sessions



DCoIP Device and System Developers

Lucent, Cisco, Nortel Networks, Ericsson, HP, Intel, DeutscheTelecom, Madge, Micom, NetManage, Netscape, Netspeak,Rockwell, Siemens, VCON, VocalTec, White Pine

Siemens - First Equipment Vendor to Provide a CompleteGlobal System Solution That Meets the Needs of New andExisting Service Providers for IP Telephony

AT&T Deployed VoIP by Installing Clarent Gateways at 38Locations (10 Domestics, and 28 Throughout the World), TwoGatekeepers - Getting Comparable Voice Quality to the PSTNand Getting Savings Up to 50 %

Netphone Services is Gaining Popularity Because Rates are 60% Lower Than Ordinary Phone Charges



DCoIP Device and System Developers (Continued)

Sprint to Trial Net2phone Services for International CallsDeploying Voice Over IP for Asian Countries to AssistCustomer Inquires in Mandarin, Cantonese and Korean

Cisco to Plan to Acquire UK-Based Internet Telephone FirmCalista ($55 Million in Stock)

Cisco to Introduce a Voice Module in 1997 for Their 3600Series Routers for Interfacing Voice, Fax, and Data AcrossExisting Data Infrastructures to Reduce Costs by RoutingPhone and Fax Traffic Over the IP Network Instead of PSTN

Analog Devices Announced First T1-on-a-Chip (ADSP-21mod980) for Data and Voice Over IP Which Can beConfigured Any Port for Any Protocol




Lucent Developed Clear Presence Audio Coder - Analternative to the ITU G.722 Wideband Speech Coding Standard (16, 24 and 32 kbps for Videoconferencing and Telephony) at 48, 56 and 64 kbps

Netspeak - WebPhone H.323 DSP Group Developed VoIP Co-processor (CT8022 DSP)

Complying With ITU-T Standards G.723.1 and G.729A+B (8 kbps)

VocalTec - Internet Phone Microsoft - MetMeeting H.323 Intel - Internet Video Phone H.323 White Pine - CU-SeeMe H.323




Nortel Networks Developed Call-Waiting to Allow Home-Based Web Surfers to Act Upon Incoming Calls Without Interrupting Their Internet Connection

Lucent, Stinger (Internet Service Provider) and Local Phone Company Can Offer DSL Service - High Speed Video, Data and Voice Sent Over Traditional Phone Lines With 30 Times Faster Than Traditional Phone Service Without Compromising the QoS

Shomiti Systems Introduced New VoIP Product, Multi QoS Parameters, Tested H.323 Family of Protocols

Selsius (Acquired by Cisco) Developed an Ethernet Telephone for Connecting to an IP-Based PBX

Touchwave Developed a PBX System Connecting to IP LAN/WAN Vienna Systems Developed an Ethernet Telephone Connecting to

the IP Network



Existing and Proposed Standards

Network ISDN PSTN Packet- B-ISDN Ethernetswitched (ATM) IEEE802.9

Multimedia H.320 H.324 H.323 H.321 H.322 Standard

Audio/voice G.711 (M) G.723.1 (M) G.711 (M) G.711 (M) G.711 (M)G.722 G.729 G.722 G.722 G.722G.728 G.728 G.728 G.728

G.723.1G.729

Audio rates, 64 5.3-6.3 64 64 64Kbps 48-64 8 48-64 48-64 48-64

1616 5.3-6.3 16 16

8Video H.261 (M) H.261 (M) H.261 (M) H.261 (M) H.261 (M)

H.263Data * T.120 T.120 T.120 T.120 T.120Multiplex H.221 (M) H.223 (M) H.225.0 (M) H.221 (M) H.221 (M)Control H.242 (M) H.245 (M) H.245 (M) H.242 (M) H.242 (M)Signaling Q.931 _ H.225.0 Q.931 Q.931

(Q.931)___________________________________________________________________________________________(M) = Mandatory

* for example, Whiteboarding application



Video Compression Techniques

Type Compression (CODEC) Rate Formats Application

H.261 p x 64Kbit/s (p is in the range 1-30).

QCIF, CIF PSTN, PSDN

H.263 20-30kbps and above QCIF, CIF SQCIF, 4CIF 16CIF. SQCIF

PSTN, PSDN, Video Conferencing, Video Telephony

H.264 Less than 1 Mb/s MPEG-4 AVC Internet Protocol-based broadcast-quality video

MPEG 2

IS-13818

4 Mbps or higher Progressive coding

broadcast quality video

MPEG4

'ISO/IEC 14496'

Less than 1.15Mb/s MPEG-4 Digital television, Interactive graphics applications, Interactive multimedia



Existing and Proposed Standards (Continued)

H.323 Entities include Terminals, Gateways, Gatekeepers and Multipoint Control Unit; APIs; Object and Source Code; Version 2 to Include H.450 and H.235. H.323 Version 2 Provides the Following: Compression Schemes, Real Security Measures, Improved Signaling, QoS, and Improved Resource Management. H.323 Enlists a Number of Other Protocols for Interoperability as Follows:

G.711, G.722, G.728, G.723.1, G.729 for Codec at the Presentation Layer (OSI model) Real-Time Transport Protocol (RTP) at the Transport Layer Resource Reservation Protocol (RSVP) at the Network Layer Real-Time Transport Control Protocol (RTCP) H.225 for Standard Call Setup Sequences / Packet Synchronization H.245 Specifies Messages for Opening and Closing Channels for Media Streams, and Other Commands, Requests and Indications at the Session Layer H.261 for Video Codec for Audiovisual at P x 64 kbps H.263 for a New Codec for Video Over PSTN T.120 Series of Multimedia Communications Protocol T.38 for Real-Time Fax; Procedures for Real Time Group 3 Facsimile Communication Between Terminals Using IP Networks

H.323 Protocol Stack



IP Protocols H.323, SIP, MGCP, Megaco/H.248

• H.323– IP communications protocol for real-time voice and video over IP.– Includes core protocol and gatekeeper toolkits.– International Telecommunications Union (ITU) recommendation for audio, video, and data

communications across IP-based networks.• SIP (Session Initiation Protocol)

– Signaling protocol for establishing real-time calls and conferences over IP networks.– SIP is an IETF (Internet Engineering Task Force) Protocol.

• MGCP (Media Gateway Control Protocol)– A complementary IETF protocol to H.323 and SIP – Defines the communication procedures for a Media Gateway Controller to provide

instructions and to gather information from Media Gateways• Megaco/H.248 (Media Gateway Control)

– Similar to MGCP, jointly defined by the IETF and ITU-T SG-16– Gradually replacing MGCP– Megaco renamed GCP (Gateway Control Protocol) -- RFC 3525



RTP / RTCP

Real-Time Transport Protocol (RTP)• Provides end-to-end delivery services of real-time Audio (G.711, G.723.1,

G.728, etc.) and Video (H.261, H.263), • Data is transported via the user datagram protocol (UDP). • RTP provides payload-type identification, sequence numbering, time

stamping, and delivery monitoring. • UDP provides multiplexing and checksum services. • RTP can be used with other transport protocols.

Real-Time Transport Control Protocol (RTCP)• Counterpart of RTP that provides control services• Primary function of RTCP is to provide feedback on the quality of the data

distribution – RTCP-XR• Carries transport-level identifier for an RTP source

– Used by receivers to synchronize audio and video.



Existing and Proposed Standards (Continued)

ITU-T Speech Coding Standards

Standards Description

G.711 64 kbps PCM (Both A-Law and u-Law) (1988) G.722 Wideband Vocoder Operating at 64, 56, or 48 kbps G.726 ADPCM Vocoder Recommendation That Folds G.721 and G.723 G.727 Embedded ADPCM Operating at 40, 32, 24, or 16 kbps G.728 16-kbps Low-Delay Code-Excited Linear Prediction Vocoder

(LD-CELP) G.729 8-kbps Conjugate-Structure Algebraic-Code-Excited Linear

Prediction (CS-ACELP) G.729A Annex A: Reduced Complexity 8 kbps CS-ACELP Speech Code G.723.1 Low-Bit-Rate Vocoder for Multimedia Communications Operating at

6.3 and 5.3 kbps (1996)



Quality of ServicesQoS

Technical Constraints

Latency is the Most Technical Problem Over Internet Telephony: by Delay, Delay Variance (or Jitter), Asymmetrical Delay, and Unpredictable Delay Twenty (20) ms Coast-to-Coast Delay in the U.S. : Mostly Not Noticeable Fifty (50) ms Delay is Noticeable 250 ms Delay by the Satellites - Conversation Becomes Difficult 350 ms Delay Over the Public Internet From Encoding and Packetizing at Both Ends of the Call Standard Half-Duplex Sound Card: Amateur Radio Conversation Quality Latency is Dependent on Lost a Packet (30 ms) or Packets, Packet Size, Buffer Size, Speaker Behavior Parameter, Protocol Application, Frame Delay, Speech Process Delay, Bridging Delay, PC Too Overloaded to Run Vocoder, and Protocol Limitations



Quality of Services (Continued)

Performance Evaluations:

Delay 200 Milliseconds From a Private IP Network With Good Encoding and Excellent DSP Technologies

Laboratory Demonstrations to Analyze Voice Quality With 100 ms, 150 ms, 200 ms, and 250 ms Latency With the Following Setups: 1. Workstation-to-Workstation Using the Gatekeeper 2. Workstation-to-Phone Using the Cisco 3620 as a H.323 Gateway

3. Phone-to-Phone Using Netrix 2210 and Cisco 3620 for Calls Connections Through IP Network



Effect of Delay on Voice Quality

PSTN

> 25ms Echo Cancellation Required

<150 ms (with echo cancellation): acceptable

150-400 ms: acceptable if delay expected



Technical Advancements

Resource ReserVation Protocol (RSVP) - It is a Receiver-Driven and up to the Receiver to Select Which Source to Receive and Amount of Bandwidth to be Reserved or Paid for Parallel IP Networks - Different Bandwidth Allocations for Data and Multimedia by Virtual or Physical Packet Networks Take on Circuit Networks Parallel or Overlay Networks are Being Built to Support Real-time Multimedia Traffic Today’s DSP Delivers More Than 15 Times the Price/Performance of Its Predecessors Ten Years Ago, Providing 100 MIPS or Faster for Voice and Video Compression and Thus Reducing Latency



VoIP Market Trends

250

106

58

23

102

135128121

115107

100

0

50

100

150

200

250

300

1996 1997 1998 1999 2000 2001

YEARS

Percent of Overall

WAN Traffic

IP

Voice

Voice vs IP GrowthSource: The Yankee Group



VoIP Market Trends

1997 19981999

20002001

20022003

0.1 0.483.6

11.7

25

47.4

81.7

0

10

20

30

40

50

60

70

80

90

Minutes (in Billions)

YEARS

Source: The Yankee Group, 1997

Voice Minutes Over IP by Year



DCoIP Market Trends

30

13

16

3

8

3

6

19

3

56

19

0

5

10

15

20

25

30

Percent of Respondents

1-5 5-10 Greater than 10

Percent of Traffic over VSANs

Internet

Intranet

Frame Relay

ATM

How Much Voice Traffic Will Migrate to VSANs Over the Next Five Years ?

Source: The Yankee Group

VSAN: Voice Services Over Alternate Networks



DCoIP Market Trends

Source: The Yankee Group

User Perceptions of Voice over the Internet

0% 50% 100%

Percentage of Respondents

No SignificantCost Savings

Lack ofStandards

Poor VoiceQuality

Not SecureEnough

Too Unreliable

Strongly Disagree

Somewhat Disagree

Neither Agree nor Disagree

Somewhat Agree

Strongly Agree



DCoIP Market Trends

More IP-Based Services - Proxy Services for H.323 More Security Features - Encryption for Conference Security, IP Security (IETF Standard) More Network Interfaces - ATM, Frame Relay, Direct Dial IP Over ISDN VoIP Market Potential - $560 Million for IT User in 1999, 34 % of Telephone Calls Carried on Packet Networks by 2005 EURO IP Telephone Market Worth $3.9 Billion by 2003 Per Dataquest Asia’s E-Commerce: Jumping to $40 Billion in 2003; Jump 100 % Annually The VoIP Market Revenues at $290 Million in 1999 and Predict Only $1.8 Billion in Sales in 2003 U.S. Unified Messaging Market for Business: $175 Million (1997 - 1998) and Lucent had 24 % Market Share Per Frost & Sullivan’s Report Real Value of VoIP is Its Ability to Integrate Voice and Data for Multimedia Applications, Not Just A Low Cost Alternate to PSTN



DCoIP Challenges

Getting Telcos up to Speed With New Technology and Willingness to Stick Around the Existing PBXs and VoIP Setting H.323, H.100 / H.110 Standards Quality of Services (QoS) Latency Problem (Delay) Advanced Voice Compression Techniques According to the Yankee Group: 83 % of Respondents Indicated That “Performance Guarantees” Are A Prime Requirement for Voice Over Alternate Networks 50 % of Respondents Indicated That “Gateway Traffic Repots” Should be Capable of Providing the Necessary Call Detail Records and Other Specified Traffic Data



References

• Voice Over IP; Uyless Black, Prentice Hall, 1999• Voice Over Data Network; Gilbert Held, McGraw-Hill, 1998• Compact PCI, Computer Telephony Specification, January, 1998• Spectral Compatibility of Digital Subscriber Line (DSL System

Tutorial, August, 2000)• Transmission Systems for Communications, Bell Telephone

Laboratories, Inc., Fourth Edition, February 1970• Communication Systems, B. P. Lathi, John Wiley & Sons, Inc.

1968• Engineering and Operations in the Bell system, Bell Telephone

Laboratories, Inc. 1978



References (Continued)

• Solutions to Problems in Modulation Theory, by Harold S. Black, Van Nostrand 1953

• How to choose the Right Dense Wavelength-Division Multiplexing (1525 to 1570 nm)

• Interactive Video Display Systems, by John P. Yu 1982• Worldwide Video Standards, Bell Telephone Laboratories, by

John P. Yu 1980• Technical Aspects of Data Communications, by John E.

McNamara, Digital Equipment Corporation 1978• Bell system Technical Reference - Switched Network

Transaction Telephone System, Pub 41804



References (Continued)

Voice Over Whatever: Internet, IP, Frame Relay, and ATM Find Their Voice; Telecommunications Planning Service, The Yankee Group, December 1997 VoIP: The Opportunity Why You Should Become An ITSP, Internet Telephony, July, 1999 Internet Technology Focus; August, September & October 1999 Lucent Website - http://www.lucent.com Cisco Website - http:// www.cisco.com Dialogic Website - http://www.dialogic.com Brooktrout Website - http://www.brooktrout.com Internet Telephony - Strategies, Solutions and Products; Andreas Wolf, ITK, AG, March, 1998 Voice Over IP Networks; Marcus Goncalves, McGraw-Hill, 1999 Delivering Voice Over IP Networks; Daniel & Emma Minoli, John Wiley & Sons, 1998

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