Week Eleven Agenda

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Week Eleven Agenda

• Attendance
• Announcements
• Review Week Ten Information
• Current Week Information
• Upcoming Assignments

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Week Eleven Topics

• Review Week Ten Information
• Interior Versus Exterior Routing Protocols
• What is convergence?
• Autonomous Systems
• Definitions
• Loop Free Path
• Current Week Information

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Interior Versus Exterior Routing Protocols

• Routing protocols designed to work inside an
  autonomous system are categorized as interior
  gateway protocols (IGPs).
• Protocols that work between autonomous systems
  are classified as exterior gateway protocols
  (EGPs).
• Protocols can be further categorized as either
  distance vector or link-state routing protocols,
  depending on their method of operation.

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Interior Versus Exterior Routing Protocols

• An interior gateway protocol (IGP) is a routing
  protocol that is used within an autonomous system
  (AS). Two types of IGP.
• Distance-vector routing protocols each router
  does not possess information about the full
  network topology. It advertises its distances to
  other routers and receives similar advertisements
  from other routers. Using these routing
  advertisements each router populates its routing
  table. In the next advertisement cycle, a router
  advertises updated information from its routing
  table. This process continues until the routing
  tables of each router converge to stable values.

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Interior Versus Exterior Routing Protocols

• Distance-vector routing protocols make routing
  decisions based on hop-by-hop. A distance vector
  routers understanding of the network is based on
  its neighbors definition of the topology, which
  could be referred to as routing by RUMOR.
• Route flapping is caused by pathological
  conditions, hardware errors, software errors,
  configuration errors, intermittent errors in
  communications links, unreliable connections
  within the network which cause certain reach
  ability information to be repeatedly advertised
  and withdrawn.

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Interior Versus Exterior Routing Protocols

• In Cisco networks, with distance vector routing
  protocols flapping routes can trigger routing
  updates with every state change.
• Cisco trigger updates are sent when these state
  changes occur. Traditionally, distance vector
  protocols do not send triggered updates.

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Interior Versus Exterior Routing Protocols

• Link-state routing protocols, each node
  possesses information about the complete network
  topology. Each node then independently calculates
  the best next hop from it for every possible
  destination in the network using local
  information of the topology. The collection of
  best next hops forms the routing table for the
  node.
• This contrasts with distance-vector routing
  protocols, which work by having each node share
  its routing table with its neighbors. In a
  link-state protocol, the only information passed
  between the nodes is information used to
  construct the connectivity maps.

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Routing Protocols

• Interior routing protocols are designed for use
  in a network that is controlled by a single
  organization
• RIPv1 RIPv2, EIGRP, OSPF and IS-IS are all
  Interior Gateway Protocols

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Link State Analogy

• Each router has a map of the network
• Each router looks at itself as the center of the
  topology
• Compare this to a you are here map at the mall
• The map is the same, but the perspective depends
  on where you are at the time

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Link State Routing Protocol

• The link-state algorithm is also known as
  Dijkstra's algorithm or as the shortest path
  first (SPF) algorithm
• The link-state routing algorithm maintains a
  complex database of topology information
• The link-state routing algorithm maintains full
  knowledge of distant routers and how they
  interconnect. They have a complete picture of the
  network

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Link State Analogy
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Distant Vector Versus Link State
Distant Vectors Routing Protocols Link State Routing Protocols
RIP (v1 and v2) OSPF
EIGRP (hybrid) IS - IS

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Exterior Gateway Routing Protocol

• An exterior routing protocol is designed for use
  between different networks that are under the
  control of different organizations
• An exterior routing routes traffic between
  autonomous systems
• These are typically used between ISPs or between
  a company and an ISP
• BGPv4is the Exterior Gateway Protocol used by all
  ISPs on the Internet

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EGI and EGP Routing Protocol
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What is Convergence

• Routers share information with each other, but
  must individually recalculate their own routing
  tables
• For individual routing tables to be accurate, all
  routers must have a common view of the network
  topology
• When all routers in a network agree on the
  topology they are considered to have converged

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Why is Quick Convergence Important?

• When routers are in the process of convergence,
  the network is susceptible to routing problems
  because some routers learn that a link is down
  while others incorrectly believe that the link is
  still up
• It is virtually impossible for all routers in a
  network to simultaneously detect a topology
  change.

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Convergence Issues

• Factors affecting the convergence time include
  the following
• Routing protocol used
• Distance of the router, or the number of hops
  from the point of change
• Number of routers in the network that use dynamic
  routing protocols
• Bandwidth and traffic load on communications
  links
• Load on the router
• Traffic patterns in relation to the topology
  change

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What are Autonomous Systems?

• An Autonomous System (AS) is a group of routers
  that share similar routing policies and operate
  within a single administrative domain.
• An AS can be a collection of routers running a
  single IGP, or it can be a collection of routers
  running different protocols all belonging to one
  organization.
• In either case, the outside world views the
  entire Autonomous System as a single entity.

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Autonomous System

• AS Numbers
• Each AS has an identifying number that is
  assigned by an Internet registry or a service
  provider.
• This number is between 1 and 65,535.
• AS numbers within the range of 64,512 through
  65,535are reserved for private use.
• This is similar to RFC 1918 IP addresses.
• Because of the finite number of available AS
  numbers, an organization must present
  justification of its need before it will be
  assigned an AS number.
• An organization will usually be a part of the AS
  of their ISP

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Autonomous System
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Autonomous System

• Each AS has its own set of rules and policies.
• The AS number uniquely distinguish it from other
  ASs around the world.

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Definitions

• Metric is a numeric value used by routing
  protocols to help determine the best path to a
  destination.
• RIP uses the metric hop count number . The lower
  the numeric value, the closer the destination.
• OSPF uses the metric bandwidth.
• EIGRP uses bandwidth

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Definitions

• Flat routing protocol is when all routing
  information is spread through the entire network.
• Hierarchical routing protocol are typically
  classless link-state protocols. This means that
  classless means that routing updates include
  subnet masks in their routing updates.
• Administrative distance is the measure used by
  Cisco routers to select the best path when there
  are two or more different routes to the same
  destination from two different routing protocols.
  Administrative distance defines the reliability
  of a routing protocol. Each routing protocol is
  prioritized in order of most to least reliable
  (believable) using an administrative distance
  value. A lower numerical value is preferred.

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Administrative Distance
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OSPF Characteristics

• OSPF is the standardized protocol for routing
  IPv4. Since its initial development, OSPF has
  been revised to be implemented with the latest
  router protocols.
• Developed for large networks (50 routers or more)
• Must be a backbone area
• Routers that operate on boundaries between the
  backbone and non-backbone are called, Area Border
  Routers (ABR)
• OSPF is a link state protocol

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OSPF Characteristics

• When the OSPF topology table is fully populated,
  the SPF algorithm calculates the shortest path to
  the destination. Triggered updates and metric
  calculation based on the cost of a specific link
  ensure quick selection of the shortest path to
  the destination.

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OSPF Characteristics
OSPF is link-state routing protocol OSPF has fast
convergence OSPF supports VLSM and CIDR
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OSPF Characteristics

• Ciscos OSPF metric is based on bandwidth
• OSPF only sends out changes when they occur.
• OSPF also uses the concept of areas to implement
  hierarchical routing
• A large internetwork can be broken up into
  multiple areas for management and route
  summarization
• OSPF is robust, but more sophisticated to
  implement.

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OSPF Characteristics
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OSPF Characteristics

• When all routers are configured into a single
  area, the convention is to use area 0(zero)
• If OSPF has more than one area, it must have an
  area 0
• Multi-area OSPF becomes more complicated to
  configure and understand
• OSPF Routing Domain
• Single Area OSPF uses only one area, usually Area
  0

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OSPF Characteristics

• 1. Flooding of link-state information
• The first thing that happens is that each node,
  router, on the network announces its own piece
  of link-state information to all other routers
  on the network. This includes who their
  neighboring routers are and the cost of the link
  between them.
• Example Hi, Im Router A, and I can reach
  Router B via a T1 link and I can reach Router C
  via an Ethernet link.
• Each router sends these announcements to all of
  the routers in the network.

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OSPF Characteristics
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OSPF Characteristics

• 2. Building a Topological Database
• Each router collects all of this link-state
  information from other routers and puts it into
  a topological database.
• 3. Shortest-Path First (SPF), Dijkstras
  Algorithm
• Using this information, the routers can
  recreate a topology graph of the network.
• Believe it or not, this is actually a very
  simple algorithm and I highly suggest you look
  at it some time, or even better, take aclass on
  algorithms.

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OSPF Characteristics

• 4. Shortest Path First Tree
• This algorithm creates an SPF tree, with the
  router making itself the root of the tree and
  the other routers and links to those routers,
  the various branches.
• 5. Routing Table
• Using this information, the router creates a
  routing table.

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OSPF Uses Areas
Hierarchical routing enables you to separate
large internetworks (autonomous systems) into
smaller internetworks that are called areas.
With this technique, routing still occurs
between the areas (called inter-area routing),
but many of the smaller internal routing
operations, such as recalculating the database
re-running the SPF algorithm, are restricted
within an area
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OSPF Uses Areas
Changes in one area are generally not propagated
(spread) to another Route summarization is
extensively used in multi-area OSPF
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OSPF Router Types
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OSPF Router Types

• Internal Routers with all their interfaces
  within the same area
• Backbone Routers with at least one interface
  connected to area 0
• ASBR(Autonomous System Boundary Router) Routers
  that have at least one interface connected to an
  external internetwork (another autonomous system)
• ABR (Area Border Router) Routers with
  interfaces attached to multiple areas.

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IS - IS Characteristics

• IS-IS is an Open System Interconnection (OSI)
  routing protocol originally specified by
  International Organization for Standardization
  (ISO)
• IS-IS is a dynamic, link-state, intra-domain,
  interior gateway protocol (IGP)
• IS-IS was designed to operate in an OSI
  Connectionless Network Service (CLNS) environment
• It was not originally designed to work with the
  IP protocol

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IS - IS Characteristics

• Extensions were added so that IS-IS can route IP
  packets
• IS-IS operates at Layer 3 (Network) of the OSI
  model
• IS-IS selects routes based upon a cost metric
  assigned to links in the IS-IS network
• A two-level hierarchy is used to support large
  routing domains
• A large domain can be administratively divided
  into areas

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OSPF and IS IS Similarities

• Classless
• Link-state databases an Dijkstras algorithm
• Hello packets to form and maintain adjacencies
• Use areas to form hierarchical topologies
• Support address summarization between areas
• Link-state representation, aging, and metrics
• Update, decision, and flooding processes
• Convergence capabilities
• Deployed on ISP backbones

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IS IS and the OSI Protocol Suite

• The OSI suite of protocols were never widely
  implemented at the Layers 3-7 because the TCP/IP
  Protocols at these layers became the de-facto
  standard.
• Layers 1 and 2 Protocols are widely used IEEE
  802.3, FDDI, IEEE 802.5, etc.

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OSI Terminology

• End system (ES) is any non-routing network node
  (host)
• Intermediate system (IS) is a router
• An area is a logical entity formed by a set of
  contiguous routers, hosts, and the data links
  that connect them
• Domain is a collection of connected areas under a
  common administrative authority(think AS)
• The areas are connected to form a backbone

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IS IS is Designed to be Hierarchical

• An OSI network is a hierarchy of these entities
• Domain -any portion of an OSI network under a
  common administration
• Area a part of a domain, broken up for easier
  management
• Backbone areas connect to other areas through
  the backbone

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IS IS is Hierarchical

• There are four levels of routing
• Level 0, routing between an ES and IS
• Level 1, routing between ISs in the same area
• Level 2, routing between different areas in the
  same domain
• Level 3, routing between separate domains

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IS IS is Hierarchical
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Why use IS IS instead of OSPF?

• IS-IS is more scalable than OSPF because it uses
  smaller LSPs for advertisements
• Up to 1000 routers can reside in an IS-IS area
  versus several hundred for OSPF
• IS-IS is more efficient with its updates and
  requires less CPU power
• IS-IS has more timers that can be fine-tuned to
  speed up convergence

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EIGRP Characteristics

• Cisco proprietary, released in 1994
• EIGRP is an advanced distance-vector routing
  protocol that relies on features commonly
  associated with link-state protocols. (sometimes
  called a hybrid routing protocol)
• Supports VLSM and CIDR
• Uses multicasts for communication not broadcasts
• Establishes adjacencies with its neighbor routers
  by using a Hello protocol
• Keeps all routes in a topology table
• Has speed and efficiency of routing updates like
  a link-state protocol

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EIGRP Metric Calculation

• By default, EIGRP uses only these
• Bandwidth (carrying capacity)
• Delay (end-to-end travel time)
• If these are the default
• Bandwidth (default)
• Delay (default)
• When are these metrics used?
• load
• Reliability
• These values are used when the administrator
  manually enters them

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EIGRP Terminology

• EIGRP uses DUAL, the Diffusing Update Algorithm
  to calculate routes not Bellman-Ford algorithm.
• The lowest cost path to a destination is called
  the feasible distance (FD)
• The cost of the route as advertised by the
  neighboring router, is called reported distance
  (RD)
• The best (primary) route to a destination is
  called the successor route (successor)
• The next best route, (backup), if there is one,
  is called the feasible successor (FS)

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EIGRP Tables

• The following three tables are maintained by
  EIGRP
• Neighbor table
• Topology table
• Routing table

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BGP
BGP is a path vector routing protocol. Defined in
RFC 1772 BGP is a distance vector routing
protocol, in that it relies on downstream
neighbors to pass along routes from their routing
table. BGP uses a list of AS numbers through
which a packet must pass to reach a destination.
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BGP Basics

• Exchange routing information between autonomous
  systems
• Guarantee the selection of a loop free path.
• BGP4 is the first version of BGP that supports
  CIDR and route aggregation.
• Common IGPs such as RIP, OSPF, and EIGRP use
  technical metrics.
• BGP does not use technical metrics.
• BGP makes routing decisions based on network
  policies, or rules (later)
• BGP does not show the details of topologies
  within each AS.
• BGP sees only a tree of autonomous systems.

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BGP Basics

• BGP updates are carried using TCP on port 179.
• In contrast, RIP updates use UDP port 520
• OSPF, IGRP, EIGRP does not use a Layer 4
  protocol
• Because BGP requires TCP, IP connectivity must
  exist between BGP peers.
• TCP connections must also be negotiated between
  them before updates can be exchanged.
• Therefore, BGP inherits those reliable,
  connection-oriented properties from TCP.

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Loop Free Path
To guarantee loop free path selection, BGP
constructs a graph of autonomous systems based on
the information exchanged between BGP neighbors.
BGP views the whole internetwork as a graph, or
tree, of autonomous systems. The connection
between any two systems forms a path. The
collection of path information is expressed as a
sequence of AS numbers called the AS Path. This
sequence forms a route to reach a specific
destination
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BGP Operation
When two routers establish a TCP-enabled BGP
connection between each other, they are called
neighbors or peers. Each router running BGP is
called a BGP speaker.
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Analog and Digital Signaling

• The human voice generates sound waves
• A telephone converts the sound waves into analog
  signals.
• However, analog transmission is not particularly
  efficient.
• The PSTN is a collection of interconnected
  voice-oriented public telephone networks, both
  commercial and government-owned.
• The PSTN today consists almost entirely of
  digital technology, except for the final link
  from the central (local) telephone office to the
  user.
• To obtain clear voice connections, the PSTN
  switches convert analog speech to a digital
  format and send it over the digital network.

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Analog and Digital Signaling

• The human voice generates sound waves
• 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 amplifiing 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 range for speech is from 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 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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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 similarbut they are not
  compatible

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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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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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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 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

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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.
• 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 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)

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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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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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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 getas in http, Sip issues an INVITE
  when someone makes a callThe 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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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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Making a Call
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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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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. The word codec is a portmanteau of
  'compressor-decompressor' or, more commonly,
  'coder-decoder.

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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
  qualityand 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

• The codecs chosen are a trade-off between
  bandwidth and voice quality
• Two techniques used to reduce voice traffic
• cRTP

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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
• cRTP compresses the header to 2 or 4 bytes
• Use on slow WAN links, but it is CPU intensive

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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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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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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 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
133
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
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Upcoming Deadlines

• Assignment 1-4-3 Data Center Design ProjectPhase
  3 Data Center Network Design is due July 11
• Assignement 10-1 Concept Questions 7 due July 4
• Assignement 11-1 Concept Question 8 due July 11
• Assignement 12-1 Concept Question 9 due July 18.