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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
Carrier
PoP
National Carrier
Backbone Network
Carrier
PoP
Carrier
PoP
Carrier
PoP
Carrier
PoP
Carrier
PoP
Metropolitan
Network
Metropolitan
Network
Carrier
PoP
Metropolitan
Network
STP
Call Establishment
DPC 246-1-2
OPC 246-1-1
CIC= 22
Initial Address
Called #
Calling #
DPC 246-1-2
OPC 246-1-1
CIC= 22
Initial Address
Called #
Calling #
Circuit
Switch
DPC 246-1-3
OPC 246-1-2
CIC= 7
Initial Address
Called #
Calling #
Circuit 22 Circuit Circuit 7
Switch
Voice Message Path
DPC 246-1-3
OPC 246-1-2
CIC= 7
Initial Address
Called #
Calling #
Circuit
Switch
Chicago
NY
San Francisco
Virginia
Carrier
POP
Carrier
POP
Carrier
POP
Managed Private/ Peer Network
ISP
Carrier
POP
ISP
ISP
ISP
Access to the Telephone Network
Access to the Telephone Network
Resi dence Customer
Central
Office
Public
and
Private
Internet
IS P
Business Customer
Carrier POP
Broadband Access to the Telephone Network
Central
Office
IP
Telephony
Adapter
PSTN
DSL
Modem
Public
and
Private
Internet
Cable
Modem
Cable Head
End Office
Cable
Modem
Cable Head
End Office
DSL
Modem
Telephone
Modem
Dial-up
Analog Line
DS LAM
Public
and
Private
Internet
Central
Office
IS P
Public
Metro
Net
Carrier POP
PSTN
IP
IP
PBX
PBX
IP Networks
Private Line, WAN,
Public Internet
IP
IP
PBX
PBX
Creating Paths for IP Packets
Across the Telephone Network
Creating a Path for Calls and Packets
MPLS Path Handler
Call Agent
Call Agent
SS7
SS7
Voice
Voice
LOCAL
PS TN
PRI
National
IP Network
MPLS Path Handler
LOCAL
PS TN
PRI
The
S tandard
Telephone Network
Standard Telephone Protocols
V.70, V.90,
V92 Dial
Terminals
H.324 PC
Terminal
H.323 V0IP Network
H.323
Gatekeeper
Server
S tandard
Telephone
IP
Network
IS DN Network
H.320
Phone
Digital
Telephone
H.323
Gateway
Router
H.323
Message
Control Unit
H.323 PC
Terminal
H.323 IP
Telephone
Multimedia Protocols
Video Equipment
Audio Equipment
User and System
Data Applications
Video Codec
H.261 and H.263
Audio Codec
G.711, G.722, G.723, G.728
G.729
T.120 and H.225
Data Transfer
H.245 Control
System Control
H.225 Call Control
H.225 RAS Control
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.
Voice Channel
Voice Signal
Output
Voltage
Frequency
(K-Hertz)
.2
1
Tone Dialing
Signals
2
3
Systems Control
Signals
4
Voice - the Most Restrictive and Smallest Tolerable Level of Delay
CB Quality
Satellite Quality
FAX and Broadcast Quality
Voice Quality
0
800
100
200
300
400
500
600
Time in msecs
150 msec Maximum
Target Voice Delay
700
Packet Switching
Packets
Q
PBX
Q
PBX
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.
The Problem of Mixing Data Packets with Voice
Large Packets “Freeze Out” Voice
Voice Packet
60 bytes
Every 20ms
Voice Packet
60 bytes
Every >214ms
Voice Packet
60 bytes
Every >214ms
~214ms Serialization Delay
Voice 1500 bytes of Data Voice
Voice 1500 bytes of Data Voice
10mbps Ethernet
Voice 1500 bytes of Data Voice
10mbps Ethernet
64kb WAN
Large packets can cause buffer filling irregularities resulting in
voice degradation
Buffers to adjust for Jitter can accommodate some delay and delay
variation
COMMON PACKET VOLUMES
TYPE
BYTE COUNT
E-mail
300B- 1500B
Client request
200B
PACKET COUNT
1
1
File transfer
50,000-500,000
30 - 300
Print submit
2,500- 25,000
2 - 18
Internet request
60 B
1
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
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
H.225
Multiplexing H.221
H.221
H.221
Signaling
Q.931
Q.931
Q.931
H.223
G.711
Q.921
H.323 PROTOCOL MULTIMEDIA OVER LANs
VIDEO
AUDIO
CONTROL/ MANAGEMENT
H.261
G.711, 722
RAS
H.263
G.723
H.225
Signaling
H.225
DATA
Control
H.245
T.124
G.728,29
RTP
UDP
X.224 Class 0
T.125
TCP
T.123
IP
LAN Layer 2
T.123
IEEE 802.3
BASICS FOR TRANSMITTING VOICE IN IP PACKETS
45 msec
Sample
Encode
64 msec
Packetize
< 100 msec
Transmit
Speaking
Network
Hearing
Output
45 msec
Decode
64 msec
Jitter
Buffer
Reconstruct
Receive
Total of Less Than 250 msecs of Delay is Tolerable
But Delay of Less than 150 msec is the Standard
Voice Quality Delay
From Various Compression Methods
Delay
Compression Method
(msec)
64K PCM (G.711)
0.75
32K ADPCM (G.726)
1
16K LD-CELP (G.728)
3–5
8K CS-ACELP (G.729)
15
8K CS-ACELP (G.729a)
15
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.
Inter-site Voice Connection Alternatives
PBX
PBX
PSTN
PBX
Gateway
Router
Local
S witch
Edge
Router
WAN
Services
Public and
Private
Internet
PBX
Gateway
Router
Edge
Router
Local
S witch
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.
General VoIP Connection Model
Telephone
C.O. Switch
Destination
IP Telephone
r
S D
m eri di a n
IP Phone
D
S
m e
r id ia n
D ua M o de
M i r oc
S
Dua Mode Mi oc
1
2 ABC 3 DEF
4I 5
L
6M
NO
7S 8
V
9
*
0Z #
Y
r
ne
a
a e
IP
Server/PBX
CISC
OSYST
EMS
WR
O
Gatekeeper
Cis co70 00
ERIES
CISC
OSYST
EMS
W
WR
O
1
2
3EF
BC D
A
4
I
5
L MN
O
7
8
S
V
*
0
Z
#
r
6
9Y
ne
a
a e
Cis co 7 0 0
ERIES
W
LAN
LAN
IP Network
LAN
LAN
Gateway
Router
Source PC Telephone
Gateway
Router
Destination PC Telephone
General Home Connection Model
Central
Office
IP
Telephony
Adapter
PSTN
DSL
Modem
Public
and
Private
Internet
Cable
Modem
Cable Head
End Office
The Standard VOIP Packet Format
VoIP
Packet
Link
IP
UDP
RTP
Voice
Header
Header
Header
Header
Payload
12 Bytes
X Bytes
X Bytes
20 Bytes
8 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.225 Call Setup
H.323
Call Proceeding
H.245
Invite
SIP
Acknowledge
SDP
CRC
MGCP
Acknowledgement
SDP
PSTN
SS7
Signaling
Network
ASSIGNING IP TELEPHONE ADDRESSES – DHCP
VOICE OVER INTERNET PROTOCOL
MODELS FOR CONNECTION
ENTERPRISE CONNECTION OVER PUBLIC AND PRIVATE NETWORKS
IP Phone
PSTN
Public Internet
IP PBX
Gateway
VOIP Router
Private
Internet
Two-Site Enterprise VOIP
IP Phone
PSTN
IP
Phone
IP PBX
Public
Internet
IP PBX
Gateway
VOIP
Router
Gateway
VOIP
Router
Private Line
or Private
Internet
Multi-Site Enterprise VOIP
IP Phone
PSTN
IP
Phone
IP PBX
Public
Internet
IP PB X
Gateway
VOIP
Router
IP Phone
Gateway
VOIP
Router
Private
Internet
IP PB X
Residence Dial over RBOC Supplied VOIP Service
Telephone
Central
Office Building
PSTN
Public
Internet
Dialup
Modem
Gateway
MultiService
VOIP
Switch
Router
Private
Internet
Residence DSL VOIP
Telephone
Central
Office Building
PSTN
DSL
Mode
m
DSLAM
Gateway
VOIP
Router
Public
Internet
Residence Cable VOIP
PSTN
Cable Head End Office
Cable
Modem
Public
Internet
CMTS
Gateway
VOIP
Router
Private
Internet
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 signaling
2. transport of the various media types
3. 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 Control
and
User Interface
Video
Video
I/O
Equipment
Audio
Audio
I/O
Equipment
Video Codec
H.261, H263
Audio Codec
G.711, G.722,
G.723,
G.723.1,
G.728, G.729
System Control
H.245
Control
Call Control
H.225. 0
RAS Control
H.225. 0
Receive Path
Delay
H.225.0 Layer
LAN Stack
User
User Data
Data
Applications
T.120
Session Layer
and Above
The Multi-layered Header of an H.323 Packet
Reliable TCP Delivery Unreliable UDP Delivery
H.245
H.225
Audio/Video Streams
Call Control RAS
TCP
RTCP and RTP
UDP
IP
Media Payload
Dialing, Compression, and Header Addressing Layers of H.323
H.323 VoIP Model
Signaling and Transport
Caller
E.164 Phone #
Audio Codec
(G.711, G729, G.723.1)
H.225, H.245, RTP, RTCP
UDP Port #
IP Address
MAC Address
802.3, ATM VPI, VCI
Frame Relay DLCI
Physical V.35, T1, DS-3
H.323 AUDIO CODING AND COMPRESSION
H.323 CALL SETUP SIGNALING AND MESSAGE FLOW
H.323 Call Setup Signaling
Admission Request
Admission Confirm
H.323
Gateway
RAS
H.323
Gateway
Setup
Connect
H.225
(Q.931)
Capabilities Exchange
Open Logical Channel
H.245
Open Logical Channel Acknowledge
Path
Resv
RTP Stream
RTP Stream
RTCP Stream
Gatekeeper
RSVP
Media
OPERATION OF
GATEWAYS AND GATEKEEPERS
IN H.323 NETWORK
Gatekeeper
Dua Mo de M i oc
RRQ, RCF
ARQ
ARQ
ACF
RRQ, RCF
H.225Call Set up
Gateway
ACF
H.225Call Set up Response
Gateway
H.245 Call Management
Real Time Message Transfer Flow
Admission-to-the-network-requests (ARQ), admission confirmation (ARC) or admission rejection (ARJ)
H.323 Call Setup and Transfer Messages through Gateway and Gatekeeper
Gatekeeper
Dua Mo de M i oc
RRQ, RCF
ARQ
ARQ
ACF
RRQ, RCF
H.225Call Set up
Gateway
ACF
H.225Call Set up Response
H.245 Call Management
Real Time Message Transfer Flow
Gateway
Three Zones Communicating by Means of Regional Area Gatekeepers
Gatekeeper 2
Gatekeeper 1
Gatekeeper 3
Dua Mo de M i oc
D uaM o de M i oc
D ua Mo de M i oc
Gateway 1
Zone 1
Gateway 2
Zone 2
Gateway 3
Zone 3
REAL TIME TRANSFER PROTOCOL
RTP
Header
V2
P
X CC M
PT
Ti mestamp
Sequence Number
Synchronization Source Identifier
UDP
Header
Source Port
Destinati on Port
Checksum
Length
Version
IHL
Type of Service
Identification
IP
Header
Ti me to Li ve
Total Length
Fl ags
Protocol
Fr agment Offset
Header Checksum
Source Address
Destinati on Address
Opti ons
Paddi ng
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.
20 bytes
IP Header
8 bytes
12 bytes
20 to 160 bytes
UDP Header
RTP Header
Voice or Media
Payload
40 byte Header
Compressed to
2 Byte Header
Compressed
Header
2 to 4 bytes
Voice or Media
Payload
20 to 160 bytes
RTP added to H.323’s Q.931 Signaling Sequence
Realtime Transfer Protocol Contribution to H.323 Q.931Signaling
H.323 Gateway
Messages
H.323 Gateway
Messages
TCP Connection
Setup
Alert
Connect H.245 Address
H.245 Gatekeeper Messages
Open Logical Channel
Open Logical Channel Acknowledge
Q.931
Signaling
Over TCP
Gatekeeper
Messages
H.245
Over TCP
Realtime Transfer Protocol Contribution
RTP Stream
RTP Stream
RTP Stream
RTCP Stream
RTP Timed Streaming
Voice Media Transfer
over UDP
VoIP Gateways and Gatekeepers Exchanging Info with the WAN
Gatekeeper
DuaM o de Mi oc
Registration
Request
Gateway
RCF
Registration
Confirm
Voice
Communication
Packets
Address
Exchange
WAN
Network
RRQ
Voice
Communication
Packets
Gateway
Universal Scheme for Connecting VoIP Traffic to Multiple Nets
National VoIP
Gatekeeper
Public and
Private
Internets
Gateway
Gatekeeper
Router
Local VOIP
Network
Gateway
H.323 Over Local
IP and Ethernet
Network
IP
Phones
Public
Circuit Switched
Telephone
Network
PBX
Standard
Analog and
Digital
Telephones
SESSION INITIATION PROTOCOL
(SIP)
FOR VOIP TRANSMISSION
Basic SIP Overall Network and Services Architecture
Application Services
SIP Servers
SIP Servers
SIP Proxy, Locate,
Register, Redirect
Processes
SIP
SIP
SIP
PSTN
SIP
RTP/UDP
PBX
IP Device w ith
SIP Agents
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
Sequence Issuance of Messages between Source and Destination Agents
SIP Server
Calling IP
Phone
Invite
Invite
Code 100 Try
SIP Signaling
(TCP)
Code 100 Try
Code 180 Ring
Code 180 Ring
Code 200 OK
Code 200 OK
ACK
Media Transfer
UDP and RTP
ACK
Media Transfer
RTP Stream
Called IP
Phone
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
Transmitted Frame with SIP, TCP, UDP and IP Headers
Gopher Kerb SMTP Telnet FTP SIP
SNMP RPC
UDP
TCP
IP
Local Area or Wide Area Net Interface
D
MAGEL A N
D
MAGEL AN
User Agent Server
User Agent Client
SIP Servers
Proxy
Redirect
Location Register
User Agent Server
User Agent Client
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
[email protected]
SIP ADDRESSING AND OPERATION
USER AGENTS USING SIP PROXY SERVERS
Initiating a SIP connection over an IP Network
Proxy Server
Invite
Redirect Server
Invite
Client
Client
IP-Based Network
User Agent
User Agent
Destination Agent Acknowledging a SIP connection over the IP Network
Proxy Server
Response Code 200
Client
Redirect Server
Response Code 200
indicating Success
Client
IP-Based Network
User Agent
User Agent
Two-way Communication with SIP connection over IP Network
Proxy
Server
ACK
User Agent
Redirect Server
ACK Acknowledgement
RTP Voice Transfer
User Agent
IP-based Network
Client
Client
FLOW USING PROXY AND REDIRECT SERVERS
SIP Signaling Sequence and Operation
SIP Server
Calling IP
Phone
Invite
Invite
Code 100 Try
SIP Signaling
(TCP )
Code 100 Try
Code 180 Ring
Code 200 OK
Code 180 Ring
Code 200 OK
ACK
Media Transfer
UDP and RTP
ACK
RTP Stream
Called IP
Phone
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.
SIP Servers and Offered Services
1. Register
“I am Frank
Groom”
Redirect
User Locations
Register
SIP
Proxy
Server
3. Locate
“Where is Kevin Groom
at 555-9999?”
5. Proxy
“I’ll establish the connection
for you”
2. Invite
“I w ant to talk to
another User Agent”
IP Devices w ith
SIP Agents
Frank’s IP Phone
4. Relocate
“Kevin is now at
[email protected]”
SIP
RTP/UDP
Kevin’s IP Phone
PBX
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.
MGCP Call Agent
MGCP
Messages
MGCP
Messages
IP NETWORK
Media
Gatway
Media
Gatway
Call Agent Signaling to IP or PSTN Network
SS7 Signal
Call Agent
PSTN
MGCP
Media
Gateway
MGCP
Call Setup
Signaling
IP
Network
Media
Gateway
Voice
Messages
Media
Gateway
Call Agent Signaling for Traditional Telephone
Transmission through IP Network
Call Agent
SS7
Signaling
Network
SS7
Signaling
Network
SS7 Signal
Trunk
Traditional
Telephone
Dial Link
Media
Gateway
MGCP
Call Setup
Signaling
IP
Network
Voice
Messages
Trunk
Media
Gateway
Traditional
Telephone
Dial Link
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
Standard
Telephone
Local
PSTN
IP Telephone
IP Telephone
Enterprise IP
Network
Enterprise IP
Network
Local
PSTN
Standard
Telephone
Carrier
SIP
Service
National IP Network
IP Telephone
Enterprise IP
Network
Local
PSTN
Standard
Telephone
Company
AT&T
SBC
Price
Service
Availability
n/a
Local and Long Distance
100 US Mkts 2004
$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 annou nced
2004
Qwest
n/a
14 State Svc Area, Local, LD
2004
Time Warner $39.95/ mo.
Partner M CI/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.
Vonage
$14.99-34.99
i2 Teleco m
$9.95/ mo.
National Bus/Res Local and LD
Local and LD
Reseller Program announced
4th qtr 2004
300,000 cust. 2004
2004
Galaxy Vo ice $19.95-34.95 Bus and Res cust, Local and LD
2004
8x8
2004
$20/ mo.
Res and Small Bus., Local, LD
Conclusion
1.Somebody has to pay.
2.Somebody has to pay taxes.
3.Somebody has to pay for
equipment.