Voice over IP
Objectives
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Circuit vs. Packet switching
Digitizing your voice
VoIP Protocols
VoIP Equipment
Cisco CallManager & Network Setup
Asterisk@Home
Pictures and demo of basic implementation
Basic Telephone Operation
 Uses circuit switching
 With circuit switching, the connection is
open for the entire phone call
 Inefficient – a lot of bandwidth is wasted
using this method
 Resources are used even when there is
silence on the line
Basic Network Operation
 Packet Switched
 With packet switching, a connection is
established only when there is data to
send
 Efficient use of bandwidth – the line does
not remain open for dead air
Voice over IP
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Voice is digitized into packets as data
These packets are then sent over the
network/internet as data would be
Packets are converted back into voice at
the receiving end
Generally, the User Datagram Protocol
(UDP) is used when dealing with VoIP
since it has less overhead than TCP
enabling faster transfers for superior voice
quality
Digitizing Your Voice
1. Using Pulse Amplitude Modulation the analog sound waves are
analyzed and sampled about 8,000 times per second
2. Relative measures of the wave strength are taken and assigned a
number in the range of 0 – 255
3. A “graph” is created which plots each sample strength against the
time it occurred
4. Using the assigned number each plot point is given an 8 bit binary
code thus turning your voice into 1’s and 0’s (Pulse Code
Modulation)
5. The process is reversed at the receiving end to convert the 1’s and
0’s back into audible voice
Digitizing Your Voice (con’t)
An analog wave is converted to digital using PAM and PCM
Features of VoIP
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VoIP supports nearly all the features of a
traditional phone system
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911 Support
Call ID
Voicemail
Music on hold
Three way calling
Protocols
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Media Gateway Control Protocol (MGCP)
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Protocol that allows the network to be seen as
a single gateway point from outside the
network
The gateway device using this protocol
provides the actual conversion between voice
and data packets
Protocols
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Session Initiation Protocol (SIP)
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Handles the interaction between VoIP clients
and VoIP servers
Specifies the systems to be used in the call
Establishes transfers and terminates the
phone calls
VoIP Equipment
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There are many vendors of VoIP equipment and many
software options available
With Cisco being one of the largest companies in the
industry the following sections will focus on
implementing VoIP using Cisco software and equipment
as well as the Asterisk@Home software from Digium Inc.
The core concepts for the implementation of each of
these systems and equipment are in essence the same
VoIP End-User equipment
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Cisco IP Telephones (Cisco 7940, 7960, IP
Softphone, Cisco 7920 Wireless Phone)
Approximate Cost: $250 - $300
Cisco 7940 IP Telephone
Cisco IP Softphone
VoIP Network Devices
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Cisco VIC-2FXO-M2 (Approximately $200)
– An expansion card that is installed into
an existing router allowing it to accept
analog lines connecting it to the PSTN and
giving it VoIP capabilities
Cisco VG 248 (Approximately $6000) – A
gateway device for interfacing analog
devices onto the network
VoIP Network Devices
Cisco VIC-2FXO-M2
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Foreign Exchange Office (FXO) - Enables an analog connection (using RJ-11) to
connect to the PSTN central office or other third party PBX system. This card DOES
NOT provide a dial tone so don’t try plugging a phone into it.
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Two-port FXO voice/fax interface card
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Generally requires version 12.0 or higher of the cisco IOS running on your device.
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The two ports on it are RJ-11 ports to provide connectivity to the PSTN and
telephony equipment.
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There are several types of expansion modules this one in particular works with
Cisco 2600, 3600, and 3700 series routers. (So potentially we could add one of
these to our Cisco routers in the lab at CCSU since we use 2600 series in our
curriculum.)
VoIP Network Devices
Cisco VG 248
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Allows up to 48 analog lines to be used as extensions
within Cisco Callmanager. Useful when not everyone needs
an IP phone, helps cut costs in switching to VoIP this way.
Supports caller ID, call waiting, conference calling, speed
dial, redial, and most other features you’d normally find on
analog phones.
Although this device only supports 48 analog lines, a Cisco
CallManager cluster can support up to 30,000 users.
VoIP Network Devices
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There are several models of switches that
can be used on the internal network
It is a good idea to purchase switches and
end-user devices that support the Powerover-Ethernet Protocol to eliminate the
need for power adapters on the end-user
devices
VoIP Network Devices
IP Phones vs. Using Analog Telephone Adapters (ATA)
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ATA’s are generally cheaper than IP phones (approx $70-100) as
opposed to $150 – 300+ for IP phones
ATA’s limit your feature set to that of a regular telephone. IP
phones are more easily integrated into the network since there is
less compatibility worries
IP phones provide features like phone books, ability to store user
settings, and offer high mobility (users can log into some IP phones
from various locations within a building or over a VPN) using an ATA
this is not possible
Home Users using services such as Vonage commonly use ATA’s or
routers with ATA’s built in like the Linksys WRT54GP2
Basic VoIP Setup of a Home User
Image taken from: http://www.quantumvoice.com/support_phone_installation.shtml
Cisco CallManager
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Cisco CallManager is installed on a server that
processes all the calls made on a Cisco VoIP
system
Normally CallManager is installed on Cisco brand
servers but other brands can be used as well
(http://www.cisco.com/en/US/products/hw/voice
app/ps378/prod_brochure_list.html)
CallManager servers are used in a clustered
fashion and one cluster can support up to
30,000 IP phones
Cisco CallManager
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Cisco CallManager servers are a key piece of a
VoIP network structure. They essentially
provide a database of the IP telephone/device
MAC addresses and map them to IP addresses
on the internal network. When an extension on
the network is dialed from the outside the
request is sent to a CallManager Server on the
network which tells the connected network
device which telephony device to ring.
Cisco CallManager
VoIP Network Diagram (Using Cisco equipment)
Asterisk@Home
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A Linux based operating system that is
essentially an IP-PBX system
Originally developed by Mark Spencer, and
employee of Digium Inc. which is the company
that markets and supports this software
A business version is also available
Provides similar core functionality to Cisco’s
CallManager
Asterisk@Home
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The company does not specify exactly how many users
the software supports
According to Digium depending upon the codec used for
voice compression anywhere from 60 – 120
simultaneous calls can be handled using a Dual 2.8 GHz
server
The business version is said to support 240+
simultaneous phone calls
An FXO Card similar to the Cisco branded card can be
added to an Asterisk server to provide the ability to
make and receive calls to and from the PSTN
Connecting
your
LAN to the
PSTN Using
Asterisk@Home
A Basic Phone Call using Session
Initiation Protocol
The server looks at
The server sends the Phone B and A exchange
the extension that is
INVITE message toWhen
ACK’sthe
(acknowledgements)
conversation hasPhone A sends an
being dialed (by
Phone B ended
and
and
anPhone
RTP session
A hangsisup
INVITE message to
Phone A) and
a BYE
established
message
tois
stream
sent and
the the SIP server
matches it with its
communication
voice between
is terminated
them and
respective IP address
communication begins
(which is the IP of
Phone B)
INVITE
BYE
ACK
A
B
Conference Calling with VoIP
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The Asterisk (and Cisco) system includes a
feature called MeetMe
MeetMe is essentially what we know as
Conference Calling
Monitoring the Asterisk Server
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The Asterisk Software supports call logs, and
monthly and daily traffic reports “out of the box”
There is an add-on package called Asterisk Stat2.0 that can be installed for additional call
monitoring functionality and reporting
The following information can be provided:
duration of calls, call destinations, call sources,
caller ID information, call date, and call time
information
Monitoring Voice Packets
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Anything from voice packets to voice quality can be
analyzed and measured using various tools
A company out of the UK named Sygnus Data
provides several different monitoring options for
VoIP network monitoring
Their various VoIP analysis tools are listed here:
http://www.sygnusdata.co.uk/3_tech.htm#10voip
Why Switch to VoIP?
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Cost Savings
Initial equipment costs can be very high but
overtime the return-on-investment (ROI)
becomes very high
 The following link provides a spreadsheet for
organizations to enter their current phone
usages and expenses for installing a VoIP
system and see the time period for their ROI:
http://www.pcmech.com/images/VoIP_Savings_
Calculator.xls
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Why Switch to VoIP?
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Mobile and Versatile
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Due to the fact that each end-user device
(e.g. phones) register to a common server an
individual can move anywhere in a building or
campus and keep their phone number and be
available almost anywhere
Using your VoIP system with a VPN can
provide remote communication and even
more cost savings from nearly any location
Our VoIP Implementation
1. Insert the Asterisk@Home Operating System CD and
start the computer
Our VoIP Implementation
2. When prompted hit the “enter” key to begin OS installation. While
waiting for the installation screen begin installing MS Windows
Server 2003 on the other two PC’s
Our VoIP Implementation
3. After Asterisk@Home installation is complete reboot and
select the “CentOS” option from the boot menu
Our VoIP Implementation
4. The first time the Asterisk OS runs it will compile itself on the
machine this may take awhile so take some time to install the XLite IP Softphone software on the Windows machines
Our VoIP Implementation
5. Once the OS has compiled login to the server and type
“netconfig” at the prompt to setup the servers network
interface
Our VoIP Implementation
6. We used the IP configuration shown below for our server
Our VoIP Implementation
7. Now that the Asterisk Server has been configured,
configure the client machines with IP addresses
Our VoIP Implementation
8. Connect the server and both client machines to
a network switch
Our VoIP Implementation
9. Use the “ping” command at the command prompt on the
client machines to verify connectivity with the Asterisk
server
Our VoIP Implementation
10. From the client machine, use Internet Explorer and type in the IP
of the Asterisk server in the Address bar to access the
Asterisk@Home web based management interface
Our VoIP Implementation
11. The Asterisk Management Portal (AMP) is where you
will setup most of the features asterisk has to offer in
our case we will use it to set up our extensions
Our VoIP Implementation
12. Fill in the extension information and password
and specify the phone protocol being used
Our VoIP Implementation
13. All extensions registered to the server along with their
names are shown in the Management Portal
Our VoIP Implementation
14. Now configure your Softphones with their respective
extension information and fill in the server information.
The fields circled below are the basic requirements for
a functioning setup.
Our VoIP Implementation
15. Your phone is now configured and you are ready to
make calls
Our VoIP Implementation
16. The phones have successfully been configured and are
able to communicate with one another
Expansion Possibilities
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Some possible ideas to expand upon the project
are:
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Explore the conference calling (MeetMe) feature of
the Asterisk OS
Implement the use of Hardware phones from various
manufacturers
Install and experiment with the monitoring
capabilities of the Asterisk-Stat 2.0 add-in package
Add the necessary hardware and server configuration
to connect our small VoIP network to the PSTN to
make and receive telephone calls
References
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Cisco IP Telephony: Solution Reference Network Design. Cisco Press. 2005.
http://64.233.161.104/search?q=cache:7ZpNEaGwEkIJ:focus.ti.com/pdfs/vf/bband/v
op_white_paper.pdf+voice+packetization&hl=en
http://areski.net/asterisk-stat-v2/about.php
http://www.cisco.com/en/US/products/hw/gatecont/ps2250/products_data_sheet091
86a008007c9bb.html
http://www.cisco.com/en/US/products/hw/routers/ps274/products_data_sheet09186
a0080091ba1.html
http://www.cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a
00800b53c7.shtml
http://www.cisco.mn/en/US/products/hw/gatecont/ps2250/products_data_sheet0918
6a0080182d38.html
http://computer.howstuffworks.com/ip-telephony.htm
http://www.digium.com/asterisk_handbook/meetme_meetmecount.html
http://www.digium.com/index.php?menu=fxsvfxo
http://www.digium.com/index.php?menu=product_detail&category=software&produc
t=ABE&tab=details
http://www.electronics.dit.ie/staff/srooney/Output/VoIP%20Assignment%20Spring%
202002.pdf
http://www.pcmech.com/show/network/771/
http://www.privateline.com/PCS/Digiprinc.htm
http://www.protocols.com/pbook/VoIPfamily.htm
http://en.wikipedia.org/wiki/MGCP
http://en.wikipedia.org/wiki/Direct_Inward_Dialing