The Basics of

Voice over the Internet Protocol

Frank M. Groom, Ph.D.

Professor of Information and Communication Sciences

Ball State University

The Telephone Network

Metropolitan Network

Metropolitan Network

Carrier PoP

Carrier PoP

Carrier PoP

Carrier PoPCarrier

PoP

Carrier PoP

Carrier PoP

National Carrier Backbone Network

Metropolitan Network

Metropolitan Network

Circuit Switch

Circuit Switch

Voice Message Path

STP Call Establishment

Circuit Switch

Circuit 22 Circuit 7

DPC 246-1-2

OPC 246-1-1

CIC= 22

Initial Address

Called #

Calling #

DPC 246-1-3

OPC 246-1-2

CIC= 7

Initial Address

Called #

Calling #

DPC 246-1-2

OPC 246-1-1

CIC= 22

Initial Address

Called #

Calling #

DPC 246-1-3

OPC 246-1-2

CIC= 7

Initial Address

Called #

Calling #

San Francisco

Chicago NY

Virginia

ISPISP ISP

Carrier POP

Carrier POP

Carrier POP Carrier

POP

ISPManaged Private/ Peer NetworkManaged Private/ Peer Network

Access to the Telephone Network

CentralOffice

ISP Carrier POP

Public and

Private Internet

Residence Customer

Business Customer

Access to the Telephone Network

CentralOffice

IP Telephony

Adapter

DSL Modem

Cable Modem

PSTN

Public and

Private Internet

Cable Head End Office

Broadband Access to the Telephone Network

CentralOffice

DSL Modem

Cable Modem

Public Metro Net

Cable Head End Office

ISP

Dial-up Analog Line

TelephoneModem

DSLAM

Carrier POP

Public and

Private Internet

IP NetworksPrivate Line, WAN,

Public Internet

PSTN

IP PBX

IP PBX

IP PBX

IP PBX

Creating Paths for IP Packets

Across the Telephone Network

Call Agent Call Agent

MPLS Path Handler

LOCAL PSTN

LOCAL PSTN

National IP Network

SS7

Voice

PRI

MPLS Path Handler

SS7

Voice

PRI

Creating a Path for Calls and Packets

V.70, V.90, V92 Dial

Terminals

H.324 PC Terminal

Standard Telephone

H.320 Phone

Digital Telephone

H.323 GatewayRouter

H.323 Message

Control Unit

H.323 Gatekeeper

ServerH.323 PC Terminal

H.323 IP Telephone

H.323 V0IP Network

IP Network

Standard Telephone Network

ISDN Network

The Standard Telephone Protocols

Video Equipment

Audio Equipment

User and SystemData Applications

System Control

H.245 Control

H.225 Call Control

H.225 RAS Control

T.120 and H.225Data Transfer

Video CodecH.261 and H.263

Audio CodecG.711, G.722, G.723, G.728

G.729

Multimedia Protocols

Standard Bodies

1.ITU – telephone standards supports the H.323 local network and conference standard.

2.IETF- Internet group supports the browser-like approach endorsed by the telephone companies.

CHARACTERISTICS OF VOICE AND IP TRAFFIC

Voice requires a Call-Setup Message to be transmitted first to notify the receiver and an Acknowledgement Message returned.

Voice requires a regularity (minimum delay) of transmission.

Voice only needs 8 Kbps bandwidth for each call.

OutputVoltage

Frequency(K-Hertz)

1 2 3 4.2

Voice Signal

Voice Channel

Tone DialingSignals

Systems ControlSignals

Voice Quality

Satellite Quality

CB Quality

0 100 200 300 400 500 600 700 800 Time in msecs

150 msec Maximum Target Voice Delay

FAX and Broadcast Quality

Voice - the Most Restrictive and Smallest Tolerable Level of Delay

Packet Switching

LANS and Router-based networks are sized based upon the clustering of a sum of independent packets submitted in a bursty fashion.

More efficient usage of trunk bandwidth is accomplished by sharing rather than determining the correct number and speed of links and ports.

The trade-off is higher delay and delay variation due to queuing, blocking and congestion against a strategy of over-provisioning facilities.

PacketsQ Q

PBX PBX

64kb WAN

~214ms Serialization Delay~214ms Serialization Delay

10mbps Ethernet 10mbps Ethernet

Voice Packet60 bytes

Every 20ms

Voice 1500 bytes of Data Voice

Voice Packet60 bytes

Every >214ms>214ms

Voice Packet60 bytes

Every >214ms>214ms

Voice 1500 bytes of Data VoiceVoice 1500 bytes of Data Voice

Large packets can cause buffer filling irregularities resulting in voice degradation

Buffers to adjust for Jitter can accommodate some delay and delay variation

The Problem of Mixing Data Packets with VoiceLarge Packets “Freeze Out” Voice

TYPE BYTE COUNT PACKET COUNT

E-mail 300B- 1500B 1

Client request 200B 1

File transfer 50,000-500,000 30 - 300

Print submit 2,500- 25,000 2 - 18

Internet request 60 B 1

COMMON PACKET VOLUMES

Result

1.Use small packets to set up a voice transfer path.

2.Use small packets to transfer voice content (73 Bytes).

3.Prioritize voice over data.

Some Standards used by VoIP

Video Conferencing Standards

NetType ISDN ATM PSTN POTS LANS

Standard H.320 H.321 H.322 H.324 H323

Year Std 1990 1995 1995 1996 1996-98

Audio G.711 G.711 G.711 G.723.1 G.711

Codec G.722,28 G.722,28 G.722,28 G.72 G.722-29

Audio Rates 64kbps 64 kbps 64 kbps 6-8kbps 6-64kbp

Video H.261 H.261 H.261 H.261 H.261

Codec H.263 H.263 H.263 H.263

Data T.120 T.120 T.120 T.120 T.120

Control H.230,42 H.242 H.242,30 H.245 H.245

Multiplexing H.221 H.221 H.221 H.223 H.225

Signaling Q.931 Q.931 Q.931 Q.921

H.323 PROTOCOL MULTIMEDIA OVER LANs

VIDEO AUDIO CONTROL/ MANAGEMENT DATA

H.261 G.711, 722 RAS Signaling Control

H.263 G.723 H.225 H.225 H.245 T.124

G.728,29

RTP X.224 Class 0 T.125

UDP TCP T.123

IP T.123

LAN Layer 2 IEEE 802.3

BASICS FOR TRANSMITTING VOICE IN IP PACKETS

Sample Encode Packetize Transmit

45 msec 64 msec < 100 msec

45 msec 64 msec

Output Decode Jitter Reconstruct Receive

Buffer

Speaking

Hearing

Total of Less Than 250 msecs of Delay is Tolerable

But Delay of Less than 150 msec is the Standard

Network

Compression Method Delay(msec)

64K PCM (G.711) 0.75

32K ADPCM (G.726) 1

16K LD-CELP (G.728)

8K CS-ACELP (G.729) 15

8K CS-ACELP (G.729a) 15

3–5

Voice Quality Delay From Various Compression Methods

KEY PARAMETERS FOR AN ACCEPTABLE VOICE NETWORK

(1) the maximum amount of delay experienced,

(2) the amount of packets that are lost, and

(3) the amount of variation or jitter in the arrival rates.

PBX

PBX

PBX

PBX

Gateway Router

Gateway Router

Edge Router

Edge Router

Local Switch

Local Switch

PSTN

WAN Services

Public and Private Internet

Inter-site Voice Connection Alternatives

VOIP OVERHEAD AND ITS EFFECTS

Packet and cell-based networks

require an overhead for

addressing and other indicators

which adds to each packet and

comprises up to 10% of the total

packet size.

r

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Cisco7000ERIES

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CISCO YSTEMSS

LAN

Telephone C.O. Switch

Gatekeeper

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3D EF

2AB C

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D ua M o de M i r oc

Cisco 700 0ERIES

OW R W

CISCO YSTEMSS

Source PC Telephone Destination PC Telephone

DestinationIP Telephone

Gateway Router

Gateway Router

S D

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3DE F

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DuaMode Mi oc

IP Phone

IP Server/PBX

LAN

IP Network LANLAN

General VoIP Connection Model

CentralOffice

IP Telephony

Adapter

DSL Modem

Cable Modem

PSTN

Public and

Private Internet

Cable Head End Office

General Home Connection Model

The Standard VOIP Packet Format

Link

Header

IP

Header

UDP

Header

RTP

Header

Voice

Payload

VoIPPacket

X Bytes 20 Bytes 8 Bytes 12 Bytes X Bytes

G.711 Standard has a 160 Byte Voice Payload

G.729 Standard has a 20 Byte Voice Payload

G.723.1 Standard has a 24 Byte Voice Payload

H.323

SIP

MGCP

H.225 Call Setup

Call Proceeding

H.245

CRC

Acknowledgement

SDP

Invite

Acknowledge

SDP

PSTN SS7

Signaling Network

ASSIGNING IP TELEPHONE ADDRESSES – DHCP

VOICE OVER INTERNET PROTOCOL

MODELS FOR CONNECTION

IP PBX

PSTN

Public Internet

Gateway VOIP Router

IP Phone

Private Internet

ENTERPRISE CONNECTION OVER PUBLIC AND PRIVATE NETWORKS

IP PBX

PSTN

Public Internet

Gateway VOIP

Router

IP Phone

Private Line or Private Internet

IP PBX

IP Phone

Gateway VOIP

Router

Two-Site Enterprise VOIP

IP PBX

PSTN

Public Internet

Gateway VOIP

Router

IP Phone

Private Internet

IP PBX

Gateway VOIP

Router

IP Phone

IP PBX

IP Phone

Multi-Site Enterprise VOIP

Telephone Central

Office Building

PSTN

Public Internet

Gateway VOIP

Router

Private Internet

MultiService Switch

Dialup Modem

Residence Dial over RBOC Supplied VOIP Service

Telephone Central

Office Building

DSL Modem

DSLAM

PSTN

PublicInternet

Gateway VOIP Router

Residence DSL VOIP

Cable Modem

CMTS

PSTN

Public Internet

Gateway VOIP Router

Cable Head End Office

Private Internet

Residence Cable VOIP

VOICE OVER IP TERMINALS

SIP Phones

H.323 Phones

VOICE OVER IP USING THE H.323 PROTOCOL

The H.323 family of protocols covers the functions of:

1. call signaling2. transport of the various media types3. system control 4. special specifications for conferencing including both point-to-point and multipoint conferencing.

To perform these functions, the family of H.323 specifications include:

(a) the control specifications of H.245 and H.225 for

signaling and control of transmission, (b) the video specifications including those for video

compression H.261 and H.263 which are performed by a video codec and are required due to the large volume that would be transmitted if left in a raw form, and

(c) the data specification of T.120. However, the most crucial H.323 specifications for employment with voice transmission under a VoIP process are those regarding the compression of audio G.711, 722, 723,728, and 729. This compression is performed as a function of the IP handset and is termed the audio codec.

VoIP—Uses Audio Component of H.323

System ControlSystem Control

H.245 Control

Call ControlH.225. 0

RAS ControlH.225. 0

VideoVideo CodecCodecH.261, H263H.261, H263

User DataUser Data Applications

T.120

H.225.0 Layer

AudioAudio I/O

Equipment

AudioAudio CodecCodecG.711, G.722,G.711, G.722,

G.723,G.723,G.723.1,G.723.1,

G.728, G.729

Receive PathReceive PathDelayDelay

System ControlSystem Control and

User Interface

VideoVideo I/O

Equipment

Session Layer

and Above

LAN Stack

Reliable TCP Delivery Unreliable UDP Delivery

Media Payload

IP

TCP UDP

H.245 H.225 Audio/Video Streams

Call Control RAS RTCP and RTP

The Multi-layered Header of an H.323 Packet

H.323 VoIP ModelSignaling and Transport

Caller

E.164 Phone #Audio Codec

(G.711, G729, G.723.1)

H.225, H.245, RTP, RTCPUDP Port #

MAC Address802.3, ATM VPI, VCIFrame Relay DLCI

Physical V.35, T1, DS-3

IP Address

Dialing, Compression, and Header Addressing Layers of H.323

H.323 AUDIO CODING AND COMPRESSION

H.323 CALL SETUP SIGNALING AND MESSAGE FLOW

H.323 Call Setup Signaling

MediaRTP StreamRTP StreamRTCP StreamRTCP Stream

Admission Request

H.323Gateway

H.245Open Logical Channel

Gatekeeper

H.225(Q.931)

Setup

Connect

Open Logical Channel Acknowledge

Capabilities Exchange

Admission Confirm RAS

RSVPPath

Resv

RTP StreamRTP Stream

H.323Gateway

OPERATION OF

GATEWAYS AND GATEKEEPERS

IN H.323 NETWORK

D u a M o d e M i o c

H.225Call Set up

H.245 Call Management

H.225Call Set up Response

Real Time Message Transfer Flow

ARQ

RRQ, RCFACF

Gatekeeper

ACF

RRQ, RCF

ARQ

GatewayGateway

Admission-to-the-network-requests (ARQ), admission confirmation (ARC) or admission rejection (ARJ)

D u a M o d e M i o c

H.225Call Set up

H.245 Call Management

H.225Call Set up Response

Real Time Message Transfer Flow

ARQ

RRQ, RCFACF

Gatekeeper

ACF

RRQ, RCF

ARQ

GatewayGateway

H.323 Call Setup and Transfer Messages through Gateway and Gatekeeper

D u a M o d e M i o c

D u a M o d e M i o c

D u a M o d e M i o c

Zone 1Zone 2

Zone 3

Gatekeeper 1Gatekeeper 2 Gatekeeper 3

Gateway 1Gateway 2 Gateway 3

Three Zones Communicating by Means of Regional Area Gatekeepers

REAL TIME TRANSFER PROTOCOL

Version IHL Type of Service Total Length

Identification Flags Fr agment Offset

Time to Live Protocol Header Checksum

Source Address

Destination Address

Options Padding

Source Port Destination Port

Length Checksum

V2 P X CC M PT Sequence Number

Timestamp

Synchronization Source Identifier

RTP Header

UDP Header

IP Header

RTP provides timing and sequencing benefits, but at the

cost of adding to the considerable UDP/IP header overhead

To assist UDP for reliability enhancement purposes, RTP

provides an additional 8-byte header to accompany the UDP

12-byte header which has already been added to the 20-byte

IP header

An additional algorithm, the RTP Compression (CRTP)

algorithm, is sometimes employed to drop the total header

to 2 bytes instead of the uncompressed 40 bytes.

.

IP Header UDP Header RTP Header Voice or MediaPayload

Compressed Voice or MediaHeader Payload

20 bytes 8 bytes 12 bytes 20 to 160 bytes

2 to 4 bytes 20 to 160 bytes

40 byte Header Compressed to 2 Byte Header

RTP Timed Streaming Voice Media Transfer

over UDPRTP StreamRTP Stream

RTCP StreamRTCP Stream

TCP ConnectionH.323 Gateway

Messages

H.245 Over TCP

Open Logical Channel

Q.931 SignalingOver TCP

Setup

Connect H.245 Address

Open Logical Channel Acknowledge

Alert

RTP StreamRTP Stream

Realtime Transfer Protocol Contribution to H.323 Q.931Signaling

H.245 Gatekeeper Messages

Realtime Transfer Protocol Contribution

H.323 GatewayMessages

Gatekeeper Messages

RTP added to H.323’s Q.931 Signaling Sequence

D u a M o d e Mi o c

Gatekeeper

GatewayGateway

WAN Network

Registration Request

Registration Confirm RRQ

RCF

Address Exchange

Voice Communication Packets

Voice Communication Packets

VoIP Gateways and Gatekeepers Exchanging Info with the WAN

H.323 Over Local IP and Ethernet

Network

Public and Private

Internets

Public Circuit Switched

Telephone NetworkGateway

Router

Standard Analog and

Digital Telephones

IP Phones

Gatekeeper

PBX

Gateway

Gatekeeper

Local VOIP Network

National VoIP

Universal Scheme for Connecting VoIP Traffic to Multiple Nets

SESSION INITIATION PROTOCOL (SIP)

FOR VOIP TRANSMISSION

Application Services

SIP Servers SIP Servers

IP Device with SIP Agents

PBX

SIP

RTP/UDP

PSTN

SIP Proxy, Locate, Register, Redirect

Processes

SIP

SIPSIP

Basic SIP Overall Network and Services Architecture

SIP MESSAGES

Messages exchanged by client phones with servers &destination phones

1. Register- Each phone must register its existence, its parameters, and it’s burned-in MAC address (likely an Ethernet address). It is then assigned a telephone number and an IP address.2. Invite- Each phone invites another phone to join a session and to exchange a conversation.3. Acknowledge- Each phone receives back an acknowledgement when a calling session has been established and the destination device agrees to converse.4. Bye- Each device issues a Bye message to hang-up and takes down the conversation session.5. Cancel- Bother servers and user devices can issue a cancellation message to stop a request in progress

InviteInvite

Code 100 TryCode 100 Try

Code 180 Ring Code 180 Ring

Code 200 OKCode 200 OK

ACK ACK

SIP Server

Calling IP Phone

Called IP Phone

SIP Signaling (TCP)

Media Transfer UDP and RTP

Media TransferRTP Stream

Sequence Issuance of Messages between Source and Destination Agents

SIP has four headers

- one used for Requests, one for Responses,

plus an Entity Header and a General Header.

- the Request header field modifies the request

command.

- the Response header field enables servers to

send response information back to the

requester.

- the Entity header field indicates information

about the message in the body of the transmitted

request/response message

Local Area or Wide Area Net Interface

IP

TCP UDP

Gopher Kerb SMTP Telnet FTP SIP SNMP RPC

Transmitted Frame with SIP, TCP, UDP and IP Headers

User Agent Client

User Agent Server

SIP ServersProxyRedirectLocation Register

D

MAGEL AN

User Agent Client

User Agent Server

D

MAGEL A N

SIP spec covers the 3 core components of VoIP system.

a) SIP first covers the application-level user agent and a

server agent that can act on behalf or the user agent and

receive and respond to the user agent requests. These

agents exist in IP phones, IP servers, and gateway

devices.

b) SIP then specifies three types of network servers that act

on behalf of clients to initiate, change, and terminate

sessions. These are the Proxy servers, Redirecting

servers, and Location and Registration servers

c) SIP also provides for addressing in the traditional Internet

URL fashion, such as with aliases of the fashion: abjones @bsu.edu or

physical addresses such 55555555555@182.182.282.182

SIP ADDRESSING AND OPERATION

USER AGENTS USING SIP PROXY SERVERS

Client

Client

User AgentUser Agent

Redirect ServerProxy Server

Invite

IP-Based Network

Invite

Initiating a SIP connection over an IP Network

Client

Client

User AgentUser Agent

Redirect ServerProxy Server

IP-Based Network

Response Code 200indicating SuccessResponse Code 200

Destination Agent Acknowledging a SIP connection over the IP Network

ClientClient

User Agent User Agent

Redirect ServerProxy Server

ACK

IP-based Network

ACK Acknowledgement

RTP Voice Transfer

Two-way Communication with SIP connection over IP Network

FLOW USING PROXY AND REDIRECT SERVERS

RTP Stream

InviteInvite

Code 100 TryCode 100 Try

Code 180 Ring Code 180 Ring

Code 200 OKCode 200 OK

ACK ACK

SIP Server

Calling IP Phone

Called IP Phone

SIP Signaling (TCP)

Media Transfer UDP and RTP

SIP Signaling Sequence and Operation

Sequence of SIP steps performed:

1. Registering with the Registration Server,

2. Requesting with an “Invite” of the proxy Server

3. Which then the Proxy Server asks the Location Server

where the desired individual is located.

4. The Redirect Server then informs the sending

requester at which URL address the desired

individual is now located.

5. The Proxy server can now make a connection to the

destination user for the caller, substituting a

national public IP address for the private local IP

address used by many businesses.

IP Devices w ith SIP Agents

PBX

SIP

RTP/UDP

SIP Servers and Offered Services

RegisterRedirectUser Locations

1. Register“I am Frank Groom”

3. Locate“Where is Kevin Groom at 555-9999?”

4. Relocate“Kevin is now at kgrm@earthlink.com”

2. Invite“I want to talk to another User Agent”

Frank’s IP Phone Kevin’s IP Phone

5. Proxy“I’ll establish the connection for you”

SIP Proxy Server

GATEWAYS AND GATEKEEPER PROTOCOLS

For simple Voice over IP connections,

H.323 or SIP protocols are satisfactory

MEDIA GATEWAY CONTROL PROTOCOL- MGCP

The MGCP protocol supports gateways between a variety of networks. Among these connections are:

a) Gateways for trunk connections between telephone networks and IP

networks.

b) Gateways interconnecting telephone networks and ATM networks.

c) Gateways from a standard PBX to a switch interface and further to a

voice carrying IP network.

d) Gateways from a home devices or home network to a connection to the

Internet primarily through an Internet Service Provider (ISP).

e) Gateway from a business or residence to the public Internet by means

of an analog modem and a dial-up connection through the

Public Switched Telephone Network (PSTN).

f) MGCP provides connection, signaling, and call control over the PSTN.

g) MGCP allows for a division of the functionality with part to be provided

by the Media Gateway and other parts to be provided by a central

Media Controller placed out in the network.

h) MGCP defines a means of handling signaling and session management

for multimedia (voice, video, and data) conferencing.

The principle components of a MGCP process include:

1. the originating IP phones,

2. the Media Gateway itself,

3. and the Call Agent as a centralized

assistant placed out in the IP network.

IP NETWORK

MGCP Call Agent

Media Gatway

Media Gatway

MGCP Messages

MGCP Messages

PSTN

IPNetwork

MGCPCall Setup Signaling

MGCP

Call Agent

Media Gateway

Media Gateway

SS7 Signal

Media Gateway

Voice Messages

Call Agent Signaling to IP or PSTN Network

IPNetwork

MGCPCall Setup Signaling

Call Agent

Media Gateway

Media Gateway

SS7 Signal

Voice Messages

SS7 Signaling Network

SS7 Signaling Network

Traditional Telephone Dial Link

Traditional Telephone Dial Link

TrunkTrunk

Call Agent Signaling for Traditional Telephone Transmission through IP Network

THE MGCP COMMANDS

The MGCP protocol implements a set of commands that control the interfaces for the media gateway. These commands are structured as a set of origination commands and a required response to each command.

A Notification Request which asks the Media Gateway to look for activity arriving on a specific port from a specific end user terminal (IP Phone).The Media Gateway sends a Notify response to the Call Agent that originally made the request to inform when one of those events occur.

The Terminal’s Call Agent sends a CreateConnection command to connect to a specific port on the Media Gateway.

The Terminal can subsequently change any of the parameters of that connection with the issuance of a ModifyConnection command.

The DelectConnect can be issued either by the terminal or the Media Gateway when a connection is no longer necessary or can’t be maintained.The status of endpoints, connections, and existing calls can be monitored by the

Call Agent by issuing AuditEndpoint or AuditConnect commands.The RestartinProgress is a notification message sent to the terminal’s Call Agent indicating that the Media Gateway or a set of connected end points are being restarted or reconnected.

The Architecture of a National IP-based SIP-VoIP Service Network

Carrier SIP Service

Local PSTN

Local PSTNLocal

PSTN

Enterprise IP NetworkIP Telephone

Standard Telephone

Enterprise IP Network

Standard Telephone

IP Telephone

National IP Network

Enterprise IP Network

Standard Telephone

IP Telephone

Company Price Service Availability

AT&T n/a Local and Long Distance 100 US Mkts 2004

SBC $29-40/mo Local and Long Distance 100 cities 2004.

Verizon n/a Bus and Res DSL, Local and LD Initial offer 2004

Bell South n/a Bus. Serv ice only announced 2004

Qwest n/a 14 State Svc Area, Local, LD 2004

Time Warner $39.95/mo. Partner MCI/Sprint, local, LD begin 2004

MCI n/a TWC partner svc Announced 2004-2005

Sprint n/a TWC partner svc announced 2004-2005

Covad $35-60/mo. National Bus/Res Local and LD 4th qtr 2004

Vonage $14.99-34.99 Local and LD 300,000 cust. 2004

i2 Telecom $9.95/mo. Reseller Program announced 2004

GalaxyVoice $19.95-34.95 Bus and Res cust, Local and LD 2004

8x8 $20/mo. Res and Small Bus., Local, LD 2004

Conclusion

1.Somebody has to pay.

2.Somebody has to pay taxes.

3.Somebody has to pay for equipment.