iSCO
`
`Integralton
`
`ROBERT CAPUTO-CCEE #1332
`
`Exhibit 1021 — 001
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 001
`
`

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`
`

`

`
`
`®)
`
`°
`C1SCO
`Packetized
`Voice and
`Data
`Integration
`Kobert Caputo
`CCIE™ #13312
`gQ@RARY OF coy
`
`"Rin,
`~~
`McGraw-Hill
`New York San Francisco Washington, D.C.
`Auckland Bogoté Caracas Lisbon London
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`0-7
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`34708-9
`
`0-07-041130-1
`
`7134647-3
`
`Voice/Data
`eee
`Cs
`wo
`
`. 1
`
`726, or
`
`Exhibit 1021 — 003
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 003
`
`

`

`
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`McGraw-Hill
`A Division ofTheMcGraw-Hill Companies
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`Copyright © 2000 by The McGraw-Hill Companies,Inc. All rights reserved.
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`Exhibit 1021 — 004
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 004
`
`

`

`Contents at a Glance
`
`Chapter 1
`
`Introduction
`
`Chapter 2
`
`Packetized Voice Overview
`
`Chapter 3
`
`Voice Concepts for Data Engineers
`
`Chapter 4
`
`Quality of Service for IP Traffic
`
`Chapter 5
`
`Quality of Service for Frame Relay
`
`Chapter 6
`
`Understanding Voice Ports and Dial Peers
`
`Chapter 7
`
`Developing a Dial Plan
`
`Chapter 8
`
`Integrating It All
`
`Chapter 9
`
`Sample Configurations
`
`Chapter 10
`
`VoIP Case Study
`
`Appendix
`Index
`327
`
`319
`
`31
`
`63
`
`tit
`
`145
`
`183
`
`219
`
`261
`
`305
`
`
`Exhibit 1021 — 005
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`vii
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 005
`
`

`

`CONTENTS|
`
`Foreword_xix
`
`Acknowledgments—xxi
`
`Chapter1
`
`Introduction
`
`Topics Covered
`Telephony Basics
`Quality of Service
`Cisco Voice Concepts
`Network Planning
`Network Implementation
`Products
`
`Who Should Read This Book?
`Level of Detail
`
`Chapter 2
`
`Packetized Voice Overview
`
`Introduction
`Business Drivers
`
`BSAoOMMoOMABW—
`
`~s
`
`10
`11
`
`Basic Transport of Voice over an
`IP Network
`
`Voice-to-Data Conversion
`Raw Data to IP Conversion
`Transport
`IP-to-Data Conversion
`
`Conversion from Data Back to Voice
`Voice overIP Basics
`
`Voice Compression
`Phone Number Mapping
`Call Leg Identification
`Session Protocol
`
`Enabling Technologies
`Hardware Advances
`Software
`Protocols
`
`
`Exhibit 1021 — 006
`
`1 1
`
`1 I 1
`
`2
`
`13
`13
`13
`14
`14
`
`17
`22
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 006
`
`

`

`
`
`|
`
`|
`|
`
`‘
`
`'
`
`xX
`
`Contents
`
`Chapter 3
`
`ManagementIssues
`Availability and Uptime
`Configuration
`Dial Plans
`Call Detail Recording
`Monitoring and Diagnostics
`Performance Management
`Cisco Voice-Enabled Routers
`Cisco 2600
`Cisco 3600
`Cisco AS5300
`Cisco 2600/3600 Voice/Fax Network Module
`Cisco AS5300 Voice/Fax Carrier Card
`
`Voice Concepts for Data Engineers
`Introduction
`Analog Telephony
`Basic Telephone Technology
`Basic Cabling and Connectors
`Voice Signal
`Signaling
`Basic Call Processing Sequence
`Other Analog Devices
`Interface Types and Signaling Methods
`Digital Voice
`Transporting Voice
`Digital Voice Signaling
`Voice Compression
`Compression Algorithms
`Comparing the Voice Encoding Techniques
`Factors That Affect Voice Quality
`Compression
`Voice Activity Detection
`Echo
`Delay
`Jitter
`Lost Packets
`Tandem Switching
`
`25
`26
`26
`27
`27
`27
`28
`28
`29
`29
`29
`30
`30
`
`31
`32
`32
`32
`32
`34
`34
`38
`4]
`42
`46
`46
`50
`52
`53
`54
`55
`56
`56
`56
`59
`60
`61
`6]
`
`Exhibit 1021 — 007
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 007
`
`

`

`Fr
`
`Contents
`
`Contents
`
`xi
`
`25
`26
`26
`27
`27
`2/7
`
`28
`28
`
`29
`29
`29
`
`30
`30
`
`31
`
`32
`32
`
`32
`32
`34
`34
`
`38
`4]
`42
`46
`46
`50
`
`52
`53
`54
`55
`56
`56
`56
`
`59
`60
`6]
`61
`
`Chapter 4
`
`Quality of Service for IP Traffic
`
`Overview
`(Compressed) Real-Time Transfer Protocol
`Configuring CRTP
`Frame-Relay Considerations
`Queuing
`FIFO Queuing
`Priority Queuing
`Custom Queuing
`Weighted Fair Queuing
`PPP Enhancements
`Configuring Multilink PPP and LFI
`Traffic Shaping
`Resource Reservation Protocol
`Signaling
`QoS Types and Queuing
`Implementation Guidelines
`IP Precedence
`IP Precedence with Voice Traffic
`Policy-Based Routing
`Configuring Policy-Based Routing
`Why Did | Set the Precedence Bit?
`Weighted Random Early Detection
`Operation of TCP
`Congestion Collapse
`Synchronization
`RED
`
`WRED Details
`Configuring WRED
`
`Chapter 5
`
`Quality of Service for Frame Relay
`Introduction
`Frame-Relay Quality of Service
`Class of Service
`Congestion Management
`Router-Based Frame Relay QoS Configuration
`Discard Eligible Tagging
`Generic Traffic Shaping
`Frame-Relay Traffic Shaping
`
`63
`
`64
`64
`66
`67
`70
`70
`70
`72
`74
`77
`78
`80
`84
`84
`85
`85
`92
`93
`94
`94
`99
`100
`10]
`102
`102
`104
`
`105
`106
`
`11]
`112
`112
`112
`114
`116
`117
`119
`125
`
`Exhibit 1021 — 008
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 008
`
`

`

`
`
`
`
`Contents
`
`Generic Traffic Shaping versus Frame-Relay
`Traffic Shaping
`Priority and Custom Queuing
`Configuring Priority and Custom Queuing
`Challenges for Frame Relay
`Future Alternatives
`
`Chapter 6
`
`Understanding Voice Ports and Dial Peers
`
`Introduction
`Voice Ports
`Physical Interfaces and Signaling
`Analog Voice Ports
`FXS
`
`FXO
`E&M
`Configuring Analog Voice Ports
`FXS and FXO Interfaces
`E&MInterfaces
`Additional Voice-Port Parameters
`E&M Signaling-Specific Commands
`Digital Interfaces
`Cisco Digital Interface Support
`Configuring PRI Interfaces
`Configuring T1 CAS
`Dial Peers
`
`Call Legs
`Dial Peers
`
`POTSDial Peers
`VoIP Dial Peers
`Finishing the Dial-Peer Statements
`Dial Peer Summary
`Dial-Peer Details
`Important Commands
`
`Developing a Dial Plan
`Introduction
`Issues Associated with Addressing
`Dial-Plan Organization
`Scaling Issues
`Interior/Exterior Routing
`
`
`
`
`
`
`
`Chapter 7
`
`
`
`136
`138
`139
`139
`143
`
`145
`
`146
`146
`146
`146
`146
`147
`147
`
`148
`148
`152
`156
`159
`160
`160
`161
`162
`164
`165
`165
`
`167
`169
`170
`171
`172
`172
`
`183
`
`184
`184
`
`184
`185
`188
`
`
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 — 009
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 009
`
`

`

`Fri.
`
`Contents
`
`Contents
`
`136
`138
`
`139
`139
`
`143
`
`145
`
`146
`146
`146
`
`146
`146
`147
`
`147
`148
`148
`
`152
`156
`
`159
`160
`160
`161
`162
`164
`165
`165
`167
`169
`170
`171
`172
`172
`
`183
`
`184
`184
`184
`185
`188
`
`Dial Plan Building Blocks
`Voice Ports
`Dial Peers
`
`Creating the Basic Plan
`Voice Network Diagram
`Create a Configuration Table
`Dial Plan Diagram Examples
`Integrating with the External World
`North American Numbering Plan
`Reserved Addresses
`
`Fitting In
`Corporate Networks
`Addressing Tools
`Number Expansions
`Hunt Groups
`Wildcards
`
`Chapter 8
`
`Integrating It All
`Introduction
`
`Requirements Analysis
`Information Gathering
`Voice-Traffic Flow Analysis
`OoS Determinations
`Path for Voice Calls
`
`Voice-Encoding Algorithm
`Router Models
`Processor Utilization
`
`Packet-Switching Mode
`Layer 2 WAN Transports
`Additional Protocols
`
`Existing QoS and Queuing Models
`Router Software Versions
`Dial Plan Creation
`
`Generating the Configuration
`Base Configuration
`Voice Connectivity
`Network Integration
`Security Filters
`Rerouting
`Network Address Translation
`
`192
`192
`192
`
`196
`196
`
`198
`198
`204
`
`206
`206
`207
`
`211
`212
`212
`
`214
`217
`
`219
`
`220
`220
`22)
`223
`
`227
`228
`228
`229
`232
`
`233
`233
`
`237
`241
`
`243
`244
`244
`
`245
`251
`
`258
`258
`
`259
`260
`
`
`
`Exhibit 1021 — 010
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 010
`
`

`

`
`
`Sample Configurations
`Introduction
`Simple Voice overIP Network
`Network Summary
`Basic Configurations
`Configuration Review
`Simple VoIP Network with Enhancements
`Design Changes
`Configuration Changes
`Complete Configurations
`PBX Trunking
`Design Changes
`Configuration Changes
`Multipoint Frame-Relay Configuration
`Network Summary
`Router Configurations
`Notes on the Multipoint Frame Configuration
`Additional Phone Support and Management Servers
`Design Changes
`Configuration Changes
`
`Contents
`
`261
`262
`262
`263
`263
`263
`269
`269
`270
`274
`279
`279
`280
`283
`283
`285
`297
`297
`299
`299
`
`305
`306
`306
`306
`309
`
`310
`
` Xiv
`
`Chapter 9
`
`Chapter 10
`
`VoIP Case Study
`Overview
`Connectivity Requirements
`Network Design
`Materials List
`Network Configuration/Tuning
`Parameters/Performance
`
`Appendix
`
`319
`
`Index
`
`327
`
`
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 — 011
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 011
`
`

`

`Fr|
`
`hapter Two
`
`Packetized Voice Overview
`
`11
`
`<—\__P
`Sound Waves
`
`—<—"_P
`Sound Waves
`
`reproduce the
`ormat.
`
`nother form of
`setup physical
`t) and transmit
`’t form connec-
`o. IP networks
`ms or packets.
`each datagram
`-hop toward its
`
`|
`|
`|
`
`|
`
`destination. To support the transport of digital voice data over this type
`of network, the voice over IP device must take the voice data, encapsu-
`late them into IP datagrams(packets), attach addressing information,
`and forward them into the network.
`
`Transport
`
`Intermediate nodes within the network inspect the addressing informa-
`tion attached to each of the IP datagrams and use that information to
`forward the datagram to the next hop along the pathto its destination.
`The networklinks can be any topology or access method that supports IP
`traffic.
`
`IP-to-Data Conversion
`
`The destination voice over IP device receives the IP datagram and
`processes it. In processing the datagram, the addressing and control
`information is removed so that the original raw data remain. The raw
`data are then presented to the voice decoding process.
`
`Conversion from Data Back to Voice
`
`The voice decoding process interprets the raw data generated by the
`source station and runs them throughthe decoding function. The output
`from the decoding function is an analog signal resembling the original
`voice signal received by the source station line feed.
`In summary, the transport of voice traffic over an IP network requires
`a conversion ofthe signalfrom analogto digital, packetization of the digi-
`tal voice data, transport of the packetized information through the net-
`work, de-packetization of the voice data, and conversion of the digital
`voice data back to an analogsignal. This processis depicted in Figure 2-2.
`
`Voice over IP Basics
`
`The previous example only dealt with the base functionality required to
`transport a voice signal across a simple data network. It lacked several
`
`Exhibit 1021 — 012
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 012
`
`

`

`
`
`
`
`12
`
`Chapter Two
`
`Analog
`Electrical
`Signal
`
`+V
`
`Binary Bit
`IP Datagrams
`Packet Stream
`Stream
`
`{|
`
`
`
`
`
`
`
`& Transmission Packetizer
`
`
`
`
`
`
`
`
`
`IP
`{0110} 1100
`
`
`P[TO]TOT]
`ve
`
`
`Analo
`.
`.
`cloctrical
`BinaryBit
`Packet Stream
`IP Datagrams
`Signal
`
`
`
`
`
`
`[iP [ 0110 1100 |
`0110111011001011 1100}|1110}|1071[0110
`CP [a7 1077]
`
`
`
`
`
`
`Analog
`IP Encapsulation
`
`
`
`
`& Transmission
`Packetizer
`Receiver
`to Digital
`
`
`Converter
`
`IP Data
`Network
`
`IP Encapsulation
`
`
`
`
`Analog
`
`<——|Transmitter
`to Digital
`Converter
`
`
`
`
`
`
`0110111011001011 0110]|1100 1110] 1011
`
`Figure 2-2
`
`Simplified voice overIP transport process.
`
`!
`
`
`
`functions that are required to build a simple voice over IP network.
`The most basic are voice compression, phone number mapping,call-leg
`identification (peer establishment), and a protocol for the transport of
`voice data.
`
`Voice Compression
`
`
`
`In orderto offset the overhead introduced by converting an analog signal
`to digital and transporting it over a packet switched network, compres-
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 — 013
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 013
`
`

`

` —
`
`‘hapter Two
`
`Packetized Voice Overview
`
`13
`
`
`
`sion algorithms can be used to reduce the amount of voice data that is
`transmitted over the packet switched network. Voice compression algo-
`rithms operate on the digital signal created by the initial analogto digi-
`tal conversion and reduce the amount of bandwidth consumed by the
`voice signal by a factor of up to 8:1 or more. This reduction helps com-
`pensate for the addressing information applied (IP, UDP, and RTP head-
`ers) to the individual packets and allows for more voice calls to transit
`links with limited bandwidth.
`
`Phone Number Mapping
`
`The simple VoIP example above assumed that each voice over IP device
`would only transmit and receive calls from a single node. In morerealis-
`tic environments, VoIP devices are required to dynamically form connec-
`tions with multiple devices. In the voice world, destinations are selected
`using a dial string or phone number. In the IP world, destinations are
`selected using IP addresses. To allow a VoIP router to communicate with
`an array of other VoIP routers, a method of mapping the phone numbers
`dialed by the telephone user to the IP addresses used by the routersis
`required. This mapping can be performed statically within each router,
`or dynamically using Domain Naming System (DNS). The planning of
`connections and their associated phone numbersis called a dial plan.
`
`Call Leg Identification
`
`In order to properly identify, process, and forwardvoicecalls, the router
`divides calls into discrete segments, referred to ascall legs. Once identi-
`fied, the router can apply services specific to each call leg, including ter-
`minating sessions.
`
`Session Protocol
`
`The Real Time Protocol (RTP) is a standardized way to transport voice and
`video traffic over IP networks. RTP adapts connectionlessIP traffic to sup-
`port real-time traffic through providing timing, sequencing, and other con-
`trol functions. RTP is defined by an Internet Engineering Task Force
`(IETF) standard to ensure interoperability in multivendor environments.
`
`Jatagrams
`
`1109]|
`= o =
`1410] 1014]
`
`
`
`IP Data
`Network
`
`
`
`
`FO] O77]
`
`atagrams
`
`r IP network.
`pping,call-leg
`e transport of
`
`| analog signal
`york, compres-
`
`x 1
`
`700]
`
`Exhibit 1021 — 014
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 014
`
`

`

`22
`
`Chapter Two
`
`
`
`Protocols
`
`Real-Time Protocol/Real-Time Control Protocol
`
`The Real-Time Protocol (RTP) and Real-Time Control Protocol (RTCP)
`were created to transport real-time traffic, such as video and audio, over
`the Internet and/or Internet Protocol-based networks. RTP is a session-
`layer protocol that usually runs on top of UDP/IP andis defined in RFCs
`1889 and 1890. UDP is the favored transport-layer protocol because its
`connectionless nature enablesits use in broadcast and multicast environ-
`ments. RTP addresses someof the elements required for real-timetraffic
`which are missing from UDP, including sequence numbering and time
`stamping. Sequence numbering enables the receiver to verify that data-
`gramshave beenreceivedin order and that intermediate datagrams were
`not lost. Time stamping helps ensure proper playback rates regardless of
`datagram arrival times. Additionally, a payload-type field helps the
`receiving station identify which application or process to hand off the data
`to. Given this functionality Cisco and most of the VoIP community have
`chosen to use RTPfor the transport of packetized voice in IP networks.
`RTCPcarries out the control functions for RTP streams. RTCP per-
`forms four discrete functions:
`
`1. Communication of information and statistics about the RTP
`stream to the application
`
`2. Identification of the RTP source
`
`8. Limiting of control traffic
`4, Secondary transport for small amounts of information
`
`This protocol is important in voice environmentsfor its ability to iden-
`tify RTP sources and for the statistical information it provides applica-
`tions.
`
`Resource Reservation Protocol
`
`Resource Reservation Protocol, or RSVP, is a protocol developed to allow
`applications to dynamically request specific levels of service from the net-
`work. RSVP’s application-based approach enables it to be customized
`based on specific application requirements. RSVP-enabled applications
`communicate their bandwidth and network latency requirements to
`receiving applications. The receivers then issue RSVP requeststo the net-
`work elements along the return path to the source station for these
`resources. Once a network element agrees to support a request, it is
`expected to honor that request for the duration of the reservation. In voice
`
`Exhibit 1021 —015
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 015
`
`

`

`yyi
`
`hapter Two
`
`Packetized Voice Overview
`
`23
`
`tocol (RTCP)
`id audio, over
`> is a session-
`
`fined in RFCs
`ol becauseits
`icast environ-
`
`al-timetraffic
`ring and time
`ify that data-
`tagrams were
`; regardless of
`eld helps the
`ad off the data
`nmunity have
`P networks.
`is. RTCP per-
`
`it the RTP
`
`ibility to iden-
`vides applica-
`
`loped to allow
`» from the net-
`
`ye customized
`d applications
`juirements to
`ssts to the net-
`tion for these
`request, it is
`ation. In voice
`
`over IP environments, a voice-enabled router can issue RSVP reserva-
`tions for each individual voice call and rely on the intermediate network
`elements to dynamically allocate the appropriate resources for the call.
`In addition to standard RSVP requests over packet networks, Cisco
`offers a means to map RSVP reservations to ATM SVCs called RSVP-
`ATM QoS interworking. This feature is supported by the 7500 series and
`enables the router to dynamically set up an ATM SVC with the appro-
`priate QoS parameters to support the RSVP request. In hybrid packet
`and cell environments, RSVP-ATM QoS interworking provides enhanced
`QoS integration and leverages ATM’s intrinsic QoS capabilities to sup-
`port packet-based requests.
`
`Diffserv/Committed Access Rate
`
`Cisco’s committed access rate (CAR) andthe goals of the IETF’s differen-
`tiated services (diffserv) working group are similar. The diffserv group is
`developing a standards-based approach to classifying IP traffic, setting
`bits within the IP headerto identify a specific QoS, providing preferential
`service based on QoS levels, and supporting the QoS levels throughout
`the network. Cisco’s committed access rate is an early implementation
`that provides rate control for IP traffic, interpretation and setting of IP
`precedence bits, and differentiated service throughout the network based
`upon IP precedencebits. This feature is significant in voice environments
`as it helps provide network-wideclasses of service for IP traffic.
`
`H.323
`
`H.323 is an International Telecommunications Union (ITU) standard
`which was created in 1996 and updated in 1998. It provides a foundation
`for audio, video, and data communications across a packet-based net-
`work infrastructure. H.323 provides standardsfor voice-encoding, simple
`bandwidth management, admission control, address translation, call
`control and management, andlinks to external networks. The voice over
`IP community has adopted H.323 standards in aneffort to foster the
`interoperability of equipment from multiple vendors. H.323 provides a
`solid foundation for building multimedia networks and has becomecriti-
`cal for developing large-scale VoIP networks.
`
`G.729/G.729a
`
`G.729 and G.729a are standard for voice-encoding algorithms. These
`algorithms analyze standard PCM voice segments and use a complex
`algorithm to encode them. Once encoded, a lookup pointer for the audio
`
`Exhibit 1021 — 016
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 016
`
`

`

`|
`
`334
`
`!
`
`|
`.
`
`|
`
`\
`
`F
`
`|
`|
`
`
`
`Index
`
`Voice quality (Cont.):
`and VAD, 56
`Voice signal, 34
`Voice-encoding algorithm, 228-230,
`258
`Voice-interface cards (VICs),
`146-148, 155
`Voice-network modules (VNMs), 147
`Voice-traffic flow analysis, 223-227
`gathering data for, 223-224
`and network bandwidth require-
`ments, 227
`processing data for, 224-225
`profiling data for, 225
`and trunking requirements,
`225-227
`
`VoIP (see Voice over IP)
`VPNs(Virtual Private Networks), 96
`
`WANs:
`fault-tolerant, 260
`outsourced, 2385-237
`Weighted fair queuing (WFQ), 19,
`74-76, 99-100
`Weighted random early detection
`(WRED), 19, 100-109
`configuring, 106-109
`and congestion collapse, 102
`queuing by, 105-106
`and synchronization, 102-103
`and TCP, 101-102
`WFQ(see Weighted fair queuing)
`Wildcards, 217
`Wink start, 46, 160, 162
`WRED(see Weighted random early
`detection)
`
`X Digital Subscriber Line (xDSL), 16
`xDSL, 16
`
`VAD (see Voice activity detection)
`vad (dial-peer command), 180-181
`VICs (see Voice-interface cards)
`Virtual Private Networks (VPNs), 96
`VNMs(voice-network modules), 147
`Voice:
`conversion of data to, 11
`conversion to data, 9-10
`Voice activity detection (VAD), 56,
`272
`Voice compression, 12-13, 52-55
`algorithmsfor, 53-55
`and quality, 55-62
`Voice network diagram, 196-197
`Voice over IP (VoIP), 8-28
`call leg identification in, 13
`dial peers in, 169-170, 194-196
`enhanced simple network, configu-
`ration of, 269-278
`and hardware advances, 14—17
`managementissuesrelated to,
`25-28
`phone number mappingin, 13
`potential benefits of, 8-9
`protocols for, 13, 22-25
`simple network, configurationof,
`262-269
`software for, 17-21
`steps in, 9-11
`voice compression in, 12-13
`Voice ports, 146-164
`analog, 146-160
`configuring, 148-160, 251,
`254-255
`definition of, 146
`in dial plan, 192
`digital, 160-164
`andsignaling types, 146-148
`Voice quality, 55-62
`and compression, 55
`and delay, 59-60
`and echo, 56-58
`andjitter, 60-61
`and lost packets, 61
`and tandem switching, 61-62
`
`
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 — 017
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 017
`
`

`

`Cisco / NETWORKING
`
`$55.00 U.S.A.
`
`0 006 457 287 8
`Cisco Packetized Voice and Data Integration gives you the knowledge andinsight to quickly get up
`to speed on planning, designing, and implementing Cisco-based voice and data networks.
`Throughtheuse of real-world examples, case studies, and detailed configuration analyses, you
`will obtain the skills necessary to successfully deploy voice and data networks.
`
`This essential sourcebookcoversall topics intrinsic to the technology, including:
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`listing of voice terminology puts the latest information at yourfingertips. Cisco Packetized Voice
`andData Integrationis an ideal hands-on guide for anynetwork administrator, engineer, or
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`CCIEinterested in using Cisco products for voice and data convergence.
`
`About the Author
`
`
`
`iin twa 7
`The comprehensive Parantxconaess‘1g voice an
`
`
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`
`|
`
`Robert Caputo, CCIE #1332, is a Lead Consultant in REALTECHSystems Corporation’s Carrier/
`Service Provider group where he works with emerging service providers to plan, design, and
`build next generation multi-service networks. Heactively consults on the deployment and
`implementation of Cisco-based products and has taught several seminars for Cisco Systems on
`voice and data convergence.
`
`
`
`
`
`tl |
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`
`|
`|
`i
`9 ¢ 8 0 0 ¢ 1
`3 4 7 ¢ 7 8
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`ISBN O-07-1347?77-1
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`Cover Design: 12E Design
`Cover Photo:Dennis 0’Clair/ Tony Stone Images
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`McGraw-Hill
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`3 9 18
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`A Division ofTheMcGraw-HillCompanies
`
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 — 018
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 018
`
`

`

`
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 019
`
`

`

`
`
`NeyUnit
`
`CTT
`
`8SS
`PACKETIZED
`VOICE
`OV
`UCBC0)
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`ther
`
`pau
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 — 020
`
`Zoho Corp. and Zoho Corp. Pvt., Ltd.
`Exhibit 1021 – 020
`
`