Week Twelve Agenda

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Week Twelve Agenda

• Attendance
• Announcements
• Monday, July 25 meet in the lab for Franklin
  Live session and Mimic Simulator Lab Assignment
  4-1-3.
• Review Week Eleven Information
• Current Week Information
• Upcoming Assignments

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Week Eleven Topics

• Review Week Ten Information
• Analog to digital signaling
• PBX and PSTN
• Definitions
• Trunk capacity
• Current Week Information
• VoIP
• Codec
• WLAN

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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 dont pick up the noise levels as
  analog signals, and doesnt induce any additional
  noise when amplifying signals.
• Digital signals hold their original form better
  than analog signals over greater distances,
  regeneration, coded, and decoded translations.

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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.

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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.

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Analog and Digital Signaling
There are several steps involved in converting
an analog signal into PCM digital format, as
shown in the figure
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Analog and Digital Signaling

• Filter analog signal remove frequencies gt 4000
  hertz
• 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).
• Digitize occurs prior to transmission over the
  telephone network (PCM process)

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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.

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Analog and Digital Signaling
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Analog and Digital Signaling

• For a sine wave, we can verify that the quantized
  values at the sampling moments are 7, 9, 11, 12,
  13, 14, 14, 15, 15, 15, 14, etc. Encoding these
  values as binary numbers would result in the
  following set of nibbles 0111 (23022121120
  104217), 1001, 1011, 1100, 1101, 1110, 1110,
  1111, 1111, 1111, 1110, etc. These digital values
  could then be further processed or analyzed by a
  purpose-specific digital signal processor or
  general purpose DSP. Several Pulse Code
  Modulation streams could also be multiplexed into
  a larger aggregate data stream, generally for
  transmission of multiple streams over a single
  physical link.

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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 similarbut they are not
  compatible.

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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.

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Public Switched Telephone Network (PSTN)
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Trunk Lines
Trunk Lines carry traffic between Central
Offices Each trunk line carries many
simultaneous conversations This is accomplished
through Time Division Multiplexing
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Time Division Multiplexing
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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).
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Private Branch Exchange (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

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PBXs and PSTN Switches
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PBXs and PSTN Switches
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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?
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Trunk Line Capacity
The science of Traffic Engineering answers this
question
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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.

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Terminology

• Blocking probability
• Grade of Service (GoS)
• Erlang
• Centum Call Second (CCS)
• Busy hour
• Busy Hour Traffic (BHT)
• Call Detail Record (CDR)

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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.

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Definitions

• One Erlang equals one full hour, or 3600 seconds,
  of telephone conversation
• The busy hour is the 60-minute period in a given
  24-hour period during which the maximum total
  traffic load occurs. The busy hour is sometimes
  called the peak hour.
• The BHT, in Erlangs 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.

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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

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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

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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

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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

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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.

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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.

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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

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Final Calculation

• To determine the appropriate number of trunks
  required to transport the traffic, the next step
  is to consult the Erlang Table, 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.

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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.

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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.

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FXS and FXO
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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).

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Connecting a Traditional PBX to the PSTN
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Telephone Signaling

• In a telephony system, a signaling mechanism is
  required for establishing and disconnecting
  telephone communications.

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Three Types of Signaling Used To Make a Phone Call

• Supervision signaling Typically characterized as
  on-hook, off-hook, 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

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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.
• EM EM is a common trunk signaling technique
  used between PBXs.

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Digital Telephone Signaling

• CAS
• CCS
• DPNSS
• ISDN
• QSIG Digital Signaling standards based protocol
  to allow different vendors PBXs to communicate
• SS7 Digital Signaling -used within the PSTN for
  signaling between PSTN switches

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Traditional Voice and Data Networks
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Integrated Voice and Data Networks
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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

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VoIP or IP Telephony?

• Cisco distinguishes between the two.
• Most technical discussions dont.
• 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.

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VoIP Connection

• To setup a VoIP communication we need the do the
  following
• The Analog to Digital Converter (ADC) 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)

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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

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VoIP
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What Protocols are Involved?
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VoIP Protocols
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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

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Components of H.323
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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

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Session Initiation Protocol (SIP) Protocols

• 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

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Session Initiation Protocol (SIP) Protocols

• 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

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Session Initiation Protocol (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.

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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
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Real-Time Control Protocol (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

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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


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VoIP
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Ciscos Solution IP Telephony

• The main component of Ciscos 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
  voice-enabled router or other gateway must be
  added where calls are forwarded to the PSTN

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Ciscos IP Telephony
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Single-Site IP Telephony
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Multisite WAN with Centralized Processing Design
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Definition of CODEC

• A codec is a device or computer program capable
  of encoding and/or decoding a digital data stream
  or signal.
• A codec encodes a data stream or signal for
  transmission, storage or encryption, or decodes
  it for playback or editing. Codecs are used in
  videoconferencing, streaming media and video
  editing applications. A video camera's
  analog-to-digital converter (ADC) converts its
  analog signals into digital signals, which are
  then passed through a video compressor for
  digital transmission or storage.

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Definition of CODEC

• A receiving device then runs the signal through
  a video de-compressor, then a digital-to-analog
  converter (DAC) for analog display. The term
  codec is also used as a generic name for a video
  conferencing unit.

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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

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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

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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)
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Reducing the Amount of Voice Traffic

• Two techniques are used to reduce voice traffic
• The codecs chosen are a trade-off between
  bandwidth and voice quality.
• Real Time Header Compression (CRTP).

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Real Time Header Compression (CRTP)

• CRTP is described in RFC2508 to reduce the header
  overhead of IP/UDP/RTP datagrams by compressing
  the three headers. The IP/UDP/RTP headers are
  compressed to 2-4 bytes most of the time.
• CRTP was designed for reliable point to point
  links with short delays. It does not perform well
  over links with high rate of packet loss, packet
  reordering and long delays.

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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.
• Use on slow WAN links, but it is CPU intensive.

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Voice Activity Detection (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

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Quality of Service (QoS) for Voice

• Classify Packets
• Mark Packets
• Marked packets can be prioritized in the scheme
  of queuing
• LLQ Ciscos Low Latency Queuing is the
  recommended method for VoIP networks

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Call Admission Control (CAC)

• 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 wont
  allow new calls to go through until bandwidth is
  available
• Callers will get a busy signal or all circuits
  busy message

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CAC
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Line Fragmentation and Interleaving (LFI)
Link fragmentation and interleaving ensures that
small voice packets dont 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.
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Wrieless Technology
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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.

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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

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Ciscos Acquisitions

• Cisco acquired the company Aironet-Aironet
  manufactured enterprise-level wireless solutions
• Cisco acquired Linksys home/small office
  wireless solutions
• Cisco acquired Airespacewireless LAN
  controllers

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What is RF?

• Radio Frequency (RF) 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.

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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 signals 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.

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Phenomena Affecting RF
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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).
• Milliwatts is 10 to the minus 3.
• 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.

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WLAN Standard Summary
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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.

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Wireless LANs
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Wireless LANs

• Cisco Small Business
• WRV210 Wireless-G VPN Router with
• RangeBooster
• http//www.cisco.com/en/US/docs/routers/csbr/wrv21
  0/quick_start/guide/WRV210_QSG_78-19169.pdf

92
Wireless LAN Standard

• The IEEE 802.11 standard defines the wireless
  LAN 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.

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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

94
Wireless LAN Standard
95
Wireless LAN Standard
96
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.
• Giga hertz is 10 to the 9th.

97
IEEE 802.11n

• 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

98
IEEE 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

99
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.
• 2. 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

100
Wi-Fi Certification

• 3. 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.

101
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
102
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 less- crowded 5 GHz band.

103
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.

104
Wireless NICs

• 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.

105
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.
106
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.

107
Wireless Access Point (AP)
108
Access Points Coverage Area
109
Mobility in a LAN
110
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.

111
Upcoming Deadlines

• Assignment 1-4-3 Data Center Design ProjectPhase
  3 Data Center Network Design is due July 11,
  2011.
• Assignment 12-1 Concept Questions 9 due July 18,
  2011.
• Assignment 13-1 Concept Questions 10 due July 25,
  2011.
• Assignment 1-4-4 Final Design Document due August
  1, 2011.