Download Week_Twelve_ppt - Franklin University Computing Sciences

Week Twelve Agenda
Attendance
Announcements
December 1, lab classroom in Phillips Hall 222 has
been requested, but not official. Mimic Simulator
Lab Assignment 4-1-3
Review Week Eleven Information
Current Week Information
Upcoming Assignments
Week Eleven Topics
Review Week Ten Information
1. Analog to digital signaling
2. PBX and PSTN
3. Definitions
4. Trunk capacity
Current Week Information
1. VoIP
2. Codec
3. WLAN
Analog and Digital Signaling
• The human voice generates sound waves
• The telephone converts the sound waves into an analog signal.
• To obtain clear voice connections, the PSTN switches convert
analog speech to a digital format and send it over the digital
network.
• At the other end of the connection, the digital signal is
converted back to analog and to the normal sound waves that
the ear can hear.
• Digital signals don’t pick up the noise levels as analog signals,
and doesn’t induce any additional noise when amplifiing
signals.
• Digital signals hold their original form better than analog
signals over greater distances, regeneration, coded, and
decoded translations.
Analog and Digital Signaling
The human range for speech is approximately
400 to 4000 hertz (hz). Higher frequencies are
filtered.
Sampling is the method used on analog signals
to formalize the digitizing process. A voltage
level corresponds to the amplitude of the
signal.
Analog and Digital Signaling
Pulse Code Modulation (PCM) is a digital
representation of an analog signal where the
magnitude of the signal is sampled regularly at
uniform intervals, then quantized to a series of
symbols in a numeric (usually binary) code.
The standard code word size is 8 bits.
Analog and Digital Signaling
There are several steps involved in converting an analog
signal into PCM digital format, as shown in the figure
Companding
• Signal is compressed for more efficient
transmission, and less noise
• Two common methods:
The A-law standard is used in Europe,
Mu-law is used in North America and
Japan
• The methods are similar—but they are not
compatible
Analog and Digital Signaling
1. Filter analog signal – remove frequencies >
4000 hertz
2. Sample – rate at least twice the highest
frequency according to Nyquist Theorem.
Samples the filtered input signal at a constant
frequency using Pulse Amplitude Modulation
(PAM).
3. Digitize – occurs prior to transmission over
the telephone network (PCM process)
Analog and Digital Signaling
4. Quantization and coding – A process that
converts each analog sample value into a discrete
value to which a unique digital code word is
assigned.
5. Companding – A process in which compression
is followed by expansion; often used for noise
reduction in equipment, in which case compression
is applied before noise exposure and expansion
after exposure.
A process in which the dynamic range of a signal
is reduced for recording purposes and then
expanded to its original value for reproduction or
playback.
Companding
• A signal is compressed for more efficient
transmission, and less noise
• Two common methods:
The A-law standard is used in Europe,
Mu-law is used in North America and
Japan
• The methods are similar—but they are not
compatible
Public Switched Telephone Network
(PSTN)
• Telephones connect to a CO (Central Office)
through the local loop
• The local loop is an analog connection
• All analog signals are converted to digital at
the CO
• Except for the local loop the entire phone
system is a modern digital network
Public Switched Telephone Network (PSTN)
Trunk Lines
Trunk Lines carry traffic between Central Offices
Each trunk line carries many simultaneous conversations
This is accomplished through Time Division Multiplexing
Time Division Multiplexing
What is a Private Branch Exchange (PBX)?
PBX is a private telephone network used within a company. The
users of the PBX phone system share a number of outside lines for
making external phone calls.
A PBX connects the internal telephones within a business and also
connects them to the public switched telephone network (PSTN).
PBX Features
• A PBX is a business telephone system that provides
business features such as call hold, call transfer, call
forward, follow-me, call park, conference calls, music
on hold, call history, and voice mail.
• Most of these features are not available in traditional
PSTN switches.
• A PBX switch often connects to the PSTN through
one or more T1 digital circuits.
• A PBX supports end-to-end digital transmission,
employs PCM switching technology, and supports
both analog and digital proprietary telephones
PBX Features
Analog PBXs were phased out about twenty years ago.
Today, most all vendors manufacture digital PBXs.
PBXs and PSTN Switches
PBXs and PSTN Switches
Trunk Line Capacity
In this diagram, 7 telephones connect to the CO in
Neighborhood A and 6 connect to the CO in
Neighborhood B
How many simultaneous conversations should this trunk line carry?
Trunk Line Capacity
The science of Traffic Engineering answers this question
What is Traffic Engineering?
• Voice traffic engineering is the science of
selecting the correct number of lines and the
proper types of service to accommodate users.
• Detailed capacity planning of all network
resources should be considered to minimize
degraded voice service in integrated networks.
• We can calculate the bandwidth required to
support a number of voice calls with a given
probability that the call will go through
Terminology
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Blocking probability
Grade of Service (GoS)
Erlang
Centum Call Second (CCS)
Busy hour
Busy Hour Traffic (BHT)
Call Detail Record (CDR)
Definitions
• The blocking probability value describes the calls that
cannot be completed because insufficient lines have
been provided. For example, a blocking probability
value of 0.01 means that 1 percent of calls would be
blocked.
• GoS is the probability that a voice gateway will block
a call while attempting to allocate circuits during the
busiest hour. GoS is written as a blocking factor, Pxx,
where xx is the percentage of calls that are blocked
for a traffic system. For example, traffic facilities that
require P01 GoS define a 1 percent probability of
callers being blocked.
Definitions
• One Erlang equals one full hour, or 3600 seconds, of
telephone conversation
• The busy hour is the 60-minute period in a given 24hour period during which the maximum total traffic
load occurs. The busy hour is sometimes called the
peak hour.
• The BHT, in Erlang’s or CCSs, is the number of
hours of traffic transported across a trunk group
during the busy hour (the busiest hour of operation).
• A CDR is a record containing information about
recent system usage, such as the identities of sources
(points of origin), the identities of destinations
(endpoints), the duration of each call, etc
Trunk Capacity Calculation
• For example, one hour of conversation (one Erlang
might be ten 6-minute calls or 15 4-minute calls.
Receiving 100 calls, with an average length of 6
minutes, in one hour is equivalent to ten Erlangs
• For example, if you know from your call logger that
350 calls are made on a trunk group in the busiest
hour and that the average call duration is 180
seconds, you can calculate the BHT as follows:
• BHT = Average call duration (seconds) * calls per
hour/3600
• BHT = 180 * 350/3600
• BHT = 17.5 Erlangs
Capacity Information
• There are years of data on the number and
duration of a phone conversation
• This historical data can be used to calculate the
capacity or number of trunk lines needed in a
telephone system
• Erlang Tables are used for this calculation
What is an Erlang Table?
• Erlang tables show the amount of traffic
potential (the BHT) for specified numbers of
circuits for given probabilities of receiving a
busy signal (the GoS)
• The BHT calculation results are stated in
Erlangs
• Erlang tables combine offered traffic (the
BHT), number of circuits, and GoS in the
following traffic models:
What is an Erlang Table?
• Erlang B: This is the most common traffic model,
which is used to calculate how many lines are
required if the traffic (in Erlangs) during the busiest
hour is known. The model assumes that all blocked
calls are cleared immediately.
• Extended Erlang B: This model is similar to ErlangB,
but it takes into account the additional traffic load
caused by blocked callers who immediately try to call
again. The retry percentage can be specified.
• Erlang C: This model assumes that all blocked calls
stay in the system until they can be handled. This
model can be applied to the design of call center
staffing arrangements in which calls that cannot be
answered immediately enter a queue
What is an Erlang Table?
• Erlang C: This model assumes that all blocked
calls stay in the system until they can be
handled. This model can be applied to the
design of call center staffing arrangements in
which calls that cannot be answered
immediately enter a queue
Trunk Capacity Calculation
• The network design is based on a star topology that
connects each branch office directly to the main
office.
• There are approximately 15 people per branch office.
• The bidirectional voice and fax call volume totals
about 2.5 hours per person per day (in each branch
office).
• Approximately 20 percent of the total call volume is
between the headquarters and each branch office.
• The busy-hour loading factor is 17 percent. In other
words, the BHT is 17% of the total traffic.
• One 64-kbps circuit supports one call.
• The acceptable GoS is P05
Trunk Capacity Calculation
• 2.5 hours call volume per user per day * 15
users = 37.5 hours daily call volume per office
• 37.5 hours * 17 percent (busy-hour load) =
6.375 hours of traffic in the busy hour
• 6.375 hours * 60 minutes per hour = 382.5
minutes of traffic per busy hour
• 382.5 minutes per busy hour * 1 Erlang/60
minutes per busy hour = 6.375 Erlangs
• 6.375 Erlangs* 20 percent of traffic to
headquarters = 1.275 Erlangs volume proposed
Final Calculation
• To determine the appropriate number of trunks
required to transport the traffic, the next step is
to consult the Erlangtable, given the desired
GoS
• This organization chose a P05 GoS. Using the
1.275 Erlangsand GoS= P05, as well as the
ErlangB table:
http://www.erlang.com/calculator/erlb/
• four circuits are required for communication
between each branch office and the
headquarters office
What do the terms FXS and FXO mean?
FXS and FXO are the name of ports used by Analog
phone lines (also known as POTS -Plain Old
Telephone Service) or phones.
FXS -Foreign eXchange Subscriber interface is the port
that actually delivers the analog line to the subscriber.
In other words it is the ‘plug on the wall’ that delivers
a dial tone, battery current and ring voltage.
What do the terms FXS and FXO mean?
FXO -Foreign eXchange Office interface is the port that
receives the analog line. It is the plug on the phone or
fax machine, or the plug(s) on your analog phone
system. It delivers an on-hook/off-hook indication
(loop closure). Since the FXO port is attached to a
device, such as a fax or phone, the device is often
called the ‘FXO device’.
FXO and FXS are always paired, i.e similar to a male /
female plug.
Without a PBX, a phone is connected directly to the
FXS port provided by a telephone company
FXS and FXO
Connecting a Traditional PBX to the PSTN
• If you have a PBX, then you connect the lines
provided by the telephone company to the PBX and
then the phones to the PBX.
• Therefore, the PBX must have both FXO ports (to
connect to the FXS ports provided by the telephone
company) and FXS ports (to connect the phone or fax
devices to).
Connecting a Traditional PBX to the PSTN
Telephone Signaling
In a telephony system, a signaling mechanism is
required for establishing and disconnecting telephone
communications.
Three Types of Signaling Used To Make a
Phone Call
• Supervision signaling: Typically characterized as on-hook, offhook, and ringing, supervision signaling alerts the CO switch
to the state of the telephone on each local loop. Supervision
signaling is used, for example, to initiate a telephone call
request on a line or trunk and to hold or release an established
connection.
• Address signaling: Used to pass dialed digits (pulse or DTMF)
to a PBX or PSTN switch. These dialed digits provide the
switch with a connection path to another telephone or
customer premises equipment.
• Informational signaling: Includes dial tone, busy tone, reorder
tone, and tones indicating that a receiver is off-hook or that no
such number exists, such as those used with call progress
indicators
Analog Telephony Signaling
• Loop start: Loop start is the simplest and least
intelligent signaling protocol, and the most
common form of local-loop signaling. Only for
residential use.
• Ground start: Also called reverse battery,
ground start is a modification of loop start that
provides positive recognition of connects and
disconnects (off-hook and on-hook)., PBXs
typically use this type of signaling.
• E&M: E&M is a common trunk signaling
technique used between PBXs.
Digital Telephone Signaling
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CAS
CCS
DPNSS
ISDN
QSIG Digital Signaling –standards based
protocol to allow different vendor’s PBXs to
communicate
• SS7 Digital Signaling -used within the PSTN
for signaling between PSTN switches
Traditional Voice and Data Networks
Integrated Voice and Data Networks
Why Integrate Voice and Data Networks?
• Integrating data, voice, and video in a network
enables vendors to introduce new features
• The unified communications network model
enables distributed call routing, control, and
application functions based on industry
standards
• Enterprises can mix and match equipment
from multiple vendors and geographically
deploy these systems wherever they are
needed
• Only one network to maintain
VoIP or IP Telephony?
• Cisco distinguishes between the two
• Most technical discussions don’t
• VoIP –analog phones and/or analog PBXs are
still used, but the analog signals are converted
to IP packets with a Voice Enabled router
• IP Telephony –IP phones are used; the system
is completely IP. Specialized call processing
software replaces the PBX –this may be called
an IP PBX
VoIP Connection
To setup a VoIP communication we need the do the following:
• The ADC (Analog to Digital Converter) converts analog voice
to digital signals (bits)
• The voice data is compressed to send the fewest number of
bits while still retaining the original information (Codec)
• Voice packets are sent using a real-time protocol (typically
RTP over UDP over IP)
• We need a signaling protocol to call users: ITU-T H323 or SIP
• At the receiver we have to disassemble packets, extract data,
then convert them to analog voice signals and send them to
sound card (or phone)
• All that must be done in a real time fashion cause we cannot
waiting for too long for a vocal answer! (QOS)
VoIP Technology
• VoIP is an “Overlay” technology
• VoIP is applied on top of an IP Network
• If the IP network is not working properly VoIP
will simply be one more thing that is broken
• Make sure the IP network is working correctly
FIRST--then implement VoIP
VoIP
What Protocols are Involved?
VoIP Protocols
H.323 Protocol
• H.323 is a standard for teleconferencing that was developed by
the International Telecommunications Union (ITU).
• It supports full multimedia audio, video and data transmission
between groups of two or more participants, and it is designed
to support large networks.
• H.323 is still a very important protocol, but it has fallen out of
use for consumer VoIP products due to the fact that it is
difficult to make it work through firewalls that are designed to
protect computers running many different applications.
• It is a system best suited to large organizations that possess the
technical skills to overcome these problems.
• As a solution for a home or small office telephony system it is
best avoided
Components of H.323
Session Initiation Protocol (SIP)
• SIP (Session Initiation Protocol) is an Internet
Engineering Task Force (IETF) standard
signaling protocol for teleconferencing,
telephony, presence and event notification and
instant messaging.
• It provides a mechanism for setting up and
managing connections, but not for transporting
the audio or video data.
• It is probably now the most widely used
protocol for managing Internet telephony
SIP Protocols
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SIP-Session Initiation Protocol
MegacoH.248 -Gateway Control Protocol
MGCP-Media Gateway Control Protocol
MIMERVP over IP -Remote Voice Protocol
Over IP Specification
SAPv2-Session Announcement Protocol
SDP-Session Description Protocol
SGCP-Simple Gateway Control Protocol
Skinny-Skinny Client Control Protocol (SCCP
SIP Protocols
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Sip is the major VoIP protocol in use today
Very similar to http
Sip uses port 5060
Sip has the same Status Codes as http
Instead of a get as in http, Sip issues an INVITE when
someone makes a call.The following are SIP responses:
1xx Informational (e.g. 100 Trying, 180 Ringing)
2xx Successful (e.g. 200 OK, 202 Accepted)
3xx Redirection (e.g. 302 Moved Temporarily)
4xx Request Failure (e.g. 404 Not Found, 482 Loop Detected)
5xx Server Failure (e.g. 501 Not Implemented)
6xx Global Failure (e.g. 603 Decline
SIP VoIP System
• User agents or phones register with a SIP Proxy.
• To initiate a session, the caller (or User Agent Client) sends a
request with the SIP URL of the called party.
• If the client knows the location of the other party it can send
the request directly to their IP address; if not, the client can
send it to a locally configured SIP network server.
• The server will resolve the called user's location and send the
request to them. During the course of locating a user, one SIP
network server can proxy or redirect the call to additional
servers until it arrives at one that definitely knows the IP
address where the called user can be found.
• Once found, the request is sent to the user.
SIP VoIP System
If phone A know the location of phone B, it can call phone
B directly without going through the proxy server
Sip uses email-style addresses to identify users
RTP
• RTP is the Real-time Transport Protocol
• RTP is used by H.323 and SIP for the actual
transmission of the VoIP packets
• RTP uses UDP
• Additionally, RTCP (Real-time Control Protocol)
provides this information:
Packet Loss
Jitter
Delay
Signal Level
Call Quality Metrics
Echo Return Loss
OSI Model
ISO Model Layer
Protocol or Standard
Presentation
Applications/CODECS
Session
H.323 and SIP
Transport
RTP / UDP / TCP
Network
IP – Non QoS
Data Link
ATM, FR, PPP, Ethernet
VoIP
Cisco’s Solution IP Telephony
• The main component of Cisco’s solution is the
Cisco Unified Communications Manager:
• It is a server used for call control and
signaling,
• It replaces a PBX
• The IP phone itself performs voice-to-IP
conversion, and voice-enabled routers are not
required within the enterprise network
• If connection to the PSTN is required, a voiceenabled router or other gateway must be added
where calls are forwarded to the PSTN
Cisco’s IP Telephony
Single-Site IP Telephony
Multisite WAN with Centralized Processing
Design
Definition of CODEC
A codec is a device or computer program
capable of encoding and/or decoding a digital
data stream or signal. The word codec is a
portmanteau of 'compressor-decompressor' or,
more commonly, 'coder-decoder‘.
Voice Coding and Compression
CODEC
• A DSP (Digital Signal Processor is a hardware component that
converts the analog signal to digital format
• Codecs are software drivers that are used to encode the speech
in a compact enough form that they can be sent in real time
across a network using the bandwidth available
• Codecs are implemented within a DSP
• VoIP software or hardware may give you the option to specify
the codecs you prefer to use
• This allows you to make a choice between voice quality and
network bandwidth usage, which might be necessary if you
want to allow multiple simultaneous calls to be held using an
ordinary broadband connection
Coding and Compression Algorithm
• The different codecs provide a certain quality of
speech
• Advances in technology have greatly improved the
quality of compressed voice and have resulted in a
variety of coding and compression algorithms
• PCM: The toll quality voice expected from the PSTN.
PCM runs at 64 kbps and provides no compression,
and therefore no opportunity for bandwidth savings
• The other algorithms use compression to save
bandwidth
• Voice quality is affected
Which CODEC is most affective?
G.729 is the recommended voice codec for most WAN
networks (that do not do multiple encodings) because of its
relatively low bandwidth requirements and high mean
opinion score (MOS) (ITU-T P.800)
Reducing the Amount of Voice Traffic
• The codecs chosen are a trade-off between
bandwidth and voice quality
• Two techniques used to reduce voice traffic:
cRTP
cRTP
• Every IP packet consists of a header and the
payload (data, voice)
• Although the payload of a voice packet is
small (20 bytes when G.729 is used), the
header is 40 bytes
• cRTP compresses the header to 2 or 4 bytes
• Use on slow WAN links, but it is CPU
intensive
VAD
Voice Activity Detection
• On average, about 35 percent of calls are silence
• In traditional voice networks, all voice calls use a
fixed bandwidth of 64 kbps regardless of how much
of the conversation is speech and how much is silence
• When VoIP is used, this silence is packetized along
with the conversation.
• VAD suppresses packets of silence, so instead of
sending IP packets of silence, only IP packets of
conversation are sent
• Therefore, gateways can interleave data traffic with
actual voice conversation traffic, resulting in more
effective use of the network bandwidth
QoS for Voice
• Classify Packets
• Mark Packets
• Marked packets can be prioritized in the
scheme of queuing
• LLQ –Cisco’s Low Latency Queuing is the
recommended method for VoIP networks
CAC –Call Admission Control
• CAC protects voice traffic from being
negatively affected by other voice traffic by
keeping excess voice traffic off the network.
• If a WAN link is fully utilized with voice
traffic then adding more voice calls will
degrade all the calls
• CAC checks if the link is maximized and
won’t allow new calls to go through until
bandwidth is available
• Callers will get a busy signal or “all circuits
busy message”
CAC
LFI
Link fragmentation
and interleaving
ensures that small
voice packets don’t
get stuck behind a
large data packet
The data packets are
fragmented into
smaller packets
The voice packets can
slip in between them
because the are
initially small.
Wrieless Technology
Wireless Technologies
• MMDS = Multichannel multipoint distribution
services used for general purpose broadband
networking. United States
• LMDS = Local multipoint distribution service
used for wireless cable television (TV),
referring to premium wireless subscription TV
rather than traditional free broadcast TV or
cable TV.
Wireless Technologies
• GSM = Global system for mobile
communication is a cellular phone protocol.
Used in many part of the world.
• GPRS = General packet radio service is a radio
technology for GSM networks. Europe and
Asia. Not related to GPS
• CDMA = Code division multiple access is a
cellular phone protocol used for digital
communication. United States
Cisco’s Acquisitions
Cisco acquired the company Aironet-Aironet
manufactured enterprise-level wireless
solutions
Cisco acquired Linksys –home/small office
wireless solutions
Cisco acquired Airespace–wireless LAN
controllers
What is RF?
• Radio frequency is a term that refers to alternating
current (AC) having characteristics such that, if the
current is input to an antenna, an electromagnetic
(EM) field is generated suitable for wireless
broadcasting and/or communications.
• Frequencies of electromagnetic radiation in the range
between 10 kHz and 300 MHz.
• Many types of wireless devices make use of RF
fields. Cordless and cellular phone , radio and
television broadcast stations, satellite
communications systems, and two-way radio services
all operate in the RF spectrum.
Phenomena Affecting RF
• Reflection: Occurs when the RF signal bounces off objects
such as metal or glass surfaces.
• Refraction: Occurs when the RF signal passes through objects
such as glass surfaces and changes direction.
• Absorption: Occurs when an object, such as a wall or
furniture, absorbs the RF signal.
• Scattering: Occurs when an RF wave strikes an uneven surface
and reflects in many directions. Scattering also occurs when an
RF wave travels through a medium that consists of objects that
are much smaller than the signal’s wavelength, such as heavy
dust.
• Diffraction: Occurs when an RF wave strikes sharp edges,
such as external corners of buildings, which bend the signal.
• Multipath: Occurs when an RF signal has more than one path
between the sender and receiver. The multiple signals at the
receiver might result in a distorted, low-quality signal.
Phenomena Affecting RF
Power Consumption by WLANs
• WLANs transmit signals just as radio stations
do to reach their listeners
• The transmit power levels for WLANs are in
milliwatts (mW), whereas for radio stations the
power levels are in megawatts (MW)
• The amount of power that can be used in
WLANs is set by the FCC
• Wireless LANs operate in the unlicensed
frequency bands, which is why they operate at
very low power levels
WLAN Standard Summary
Wireless LANs
• 802.11 wireless LANs extend the 802.3 Ethernet LAN
infrastructures to provide additional connectivity options.
• In an 802.3 Ethernet LAN, each client has a cable that
connects the client NIC to a switch.
• The switch is the point where the client gains access to the
network.
• In a wireless LAN, each client uses a wireless adapter to gain
access to the network through a wireless device such as a
wireless router or access point.
• The wireless adapter in the client communicates with the
wireless router or access point using RF signals.
• Once connected to the network, wireless clients can access
network resources just as if they were wired to the network.
Wireless LANs
Wireless LAN Standard
802.11 wireless LAN is an IEEE standard that
defines how radio frequency (RF) in the
unlicensed industrial, scientific, and medical
(ISM) frequency bands is used for the physical
layer and the MAC sub-layer of wireless links.
Data Rate:
802.11: 1 -2 Mb/s data rates
802.11a and g: support up to 54 Mb/s,
802.11b: supports up to a maximum of 11
Mb/s
802.11n: Up to 500 Mb/s.
Wireless LAN Standard
Modulation technique:
Direct Sequence Spread Spectrum (DSSS)
802.11b, 802.11g
Orthogonal Frequency Division
Multiplexing (OFDM).
802.11a, 802.11g, 802.11n
Band:
2.4 GHz:
802.11b, 802.11g, 802.11n
5 GHz:
802.11a, 802.11n
Wireless LAN Standard
Wireless LAN Standard
Wireless LAN Standard
802.11a
OFDM modulation and uses the 5 GHz band.
Less likely to experience interference than devices that operate in
the 2.4 GHz band
Because there are fewer consumer devices that use the 5 GHz band.
There are some important disadvantages to using the 5 GHz band.
The first is that higher frequency radio waves are more easily
absorbed by obstacles such as walls, making 802.11a susceptible to
poor performance due to obstructions.
The second is that this higher frequency band has slightly poorer
range than either 802.11b or g.
Also, some countries, including Russia, do not permit the use of the
5 GHz band, which may continue to curtail its deployment.
802.11n
The IEEE 802.11n standard is intended to improve WLAN
data rates and range without requiring additional power or RF
band allocation.
802.11n uses multiple radios and antennas at endpoints, each
broadcasting on the same frequency to establish multiple
streams.
The multiple input/multiple output (MIMO) technology splits
a high data-rate stream into multiple lower rate streams and
broadcasts them simultaneously over the available radios and
antennae.
This allows for a theoretical maximum data rate of 248 Mb/s
using two streams.
The standard is now ratified
802.11n
Operates in the 2.4 GHz band or in the 5 GHz band
The 2.4GHz band is more crowded with interference
from lots of other devices and 802.11g networks
The 5GHz band is less crowded but the range is less
Terminology:
• A “dual-frequency” or “dual-band” AP allows you to
select which band 2.4GHz or 5 GHz
• A “dual-radio” AP allows the AP to operate at both
frequencies
• You can allows your old 802.11g clients to connect
on the 2.4 GHz band and your new 802.11n clients to
connect on the 5GHz band
Wi-Fi Certification
The 3 key organizations influencing WLAN standards are:
• ITU-R
ITU-R regulates allocation of RF bands.
The ITU-R regulates the allocation of the RF spectrum.
• IEEE
IEEE specifies how RF is modulated to carry information.
The IEEE developed and maintains the standards for local
and metropolitan area networks. The dominant
standards in the IEEE 802 are 802.3 Ethernet, and 802.11
Wireless LAN
Wi-Fi Certification
• Wi-Fi Alliance (www.wi-fi.org)
Wi-Fi ensures that vendors make devices
that are interoperable.
The Wi-Fi Alliance is to improve the
interoperability of products by certifying
vendors for conformance to industry
norms and adherence to standards.
Certification includes all three IEEE 802.11
RF technologies, as well as early adoption
of pending IEEE drafts, such as 802.11n,
and the WPA and WPA2security standards
based on IEEE 802.11i.
802.11g and n (2.4GHz)
Although there are 11 channels, these channels overlap each other
You can have only use three APs in close proximity without
interference.
The APS will operate on channels 1, 6 and 11
802.11a and n (5GHz)
• Twelve separate non-overlapping channels
• As a result, you can have up to twelve access points set to different
channels in the same area without them interfering with each other.
• This makes access point channel assignment much easier and
significantly increases the throughput the wireless LAN can deliver
within a given area.
• In addition, RF interference is much less likely because of the lesscrowded 5 GHz band.
Wireless NICs
• The device that makes a client station capable of sending and
receiving RF signals is the wireless NIC.
• Like an Ethernet NIC, the wireless NIC, using the modulation
technique it is configured to use, encodes a data stream onto
an RF signal.
• Wireless NICs are most often associated with mobile devices,
such as laptop computers.
• In the 1990s , wireless NICs for laptops were cards that
slipped into the PCMCIA slot.
• PCMCIA wireless NICs are still common, but many
manufacturers have begun building the wireless NIC right into
the laptop.
• Unlike 802.3 Ethernet interfaces built into PCs, the wireless
NIC is not visible, because there is no requirement to connect
a cable to it.
Wireless NICs
Other options have emerged over the years as well. Desktops located
in an existing, non-wired facility can have a wireless PCI NIC
installed.
To quickly set up a PC, mobile or desktop, with a wireless NIC, there
are many USB options available as well.
Wireless Access Point (AP)
• An access point connects wireless clients (or stations)
to the wired LAN.
• An access point is a Layer 2 device that functions like
an 802.3 Ethernet hub.
• Client devices do not typically communicate directly
with each other; they communicate with the AP.
• In essence, an access point converts the TCP/IP data
packets from their 802.11 frame encapsulation format
in the air to the 802.3 Ethernet frame format on the
wired Ethernet network.
Wireless Access Point (AP)
Access Point’s Coverage Area
Mobility in a LAN
Autonomous AP
• Originally in WLANs, all of the configurations
and management was done on each access
point
• This type of access point was a stand-alone
device
• The term for this is a fat AP, standalone AP,
intelligent AP, or, most commonly, an
autonomous AP
• All encryption and decryption mechanisms and
MAC layer mechanisms also operate within
the autonomous AP
Upcoming Deadlines
• Assignment 1-4-3 Data Center Design Project
Phase 3: Data Center Network Design is due
November 24,2010.
• Assignment 11-1, Concept Question 8, due
November 24, 2010
• Assignment 12-1, Concept Questions 9, due
December 1, 2010.
• Assignement 13-1 Concept Questions 10 is
due December 8, 2010.
• Assignment 1-4-4 Final Design Document is
due December 15, 2010.