Transcript OSPF - Computing Sciences

Week Eleven Agenda
Attendance
Announcements
Mimic Simulator Lab Assignment 4-1-2, Basic Routing
and LAN Switching Configuration
Review Week Ten Information
Current Week Information
Upcoming Assignments
Week Eleven Topics
Review Week Ten Information
1. Interior Versus Exterior Routing Protocols
2. What is convergence?
3. Autonomous Systems
4. Definitions
5. Loop Free Path
Current Week Information
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.
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.
Interior Versus Exterior Routing Protocols
Distance-vector routing protocols make routing
decisions based on hop-by-hop. A distance vector
router’s 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repeatedly advertised and withdrawn.
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.
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.
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
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
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
Link State Analogy
Distant Vector Versus Link State
Distant Vectors Routing Protocols
Link State Routing Protocols
RIP (v1 and v2)
OSPF
EIGRP (hybrid)
IS - IS
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
EGI and EGP Routing Protocol
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
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.
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
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.
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
Autonomous System
Autonomous System
• Each AS has its own set of rules and policies.
• The AS number uniquely distinguish it from
other ASs around the world.
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
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.
Administrative Distance
EIGRP Characteristics
EIGRP is an advanced distance vector protocol that
employs the best features of link-state routing.
OSPF Characteristics
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OSPF is the standardized protocol for routing IPv4.
Since it’s 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
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.
OSPF Characteristics
OSPF is link-state routing protocol
RIP and EIGRP are distance-vector (routing by rumor) routing
protocols, susceptible to routing loops, split-horizon, and other
issues.
OSPF has fast convergence
RIP hold-down timers can cause slow convergence.
OSPF supports VLSM and CIDR
RIPv1 does not
OSPF Characteristics
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Cisco’s OSPF metric is based on bandwidth
RIP is based on hop count
OSPF only sends out changes when they occur.
RIP sends entire routing table every 30 seconds,
IGRP every 90 seconds
• 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
OSPFCharacteristics
• Two open-standard routing protocols to choose from:
RIP, simple but very limited, or
OSPF, robust but more sophisticated to
implement.
EIGRP is Cisco proprietary
OSPF Characteristics
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
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, I’m 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.
OSPF Characteristics
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.
Large OSPF Networks
Large link-state table
Each router maintains a LSDB for all links in the area
The LSDB requires the use of memory
Frequent SPF calculations
A topology change in an area causes each router to
re-run SPF to rebuild the SPF tree and the routing
table.
A flapping link will affect an entire area.
SPF re-calculations are done only for changes within
that area.
Issues with large OSPFNetworks
Large routing table
Typically, the larger the area the larger the routing
table.
A larger routing table requires more memory and
takes more time to perform the route look-ups.
Solution: Divide the network into multiple areas
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 interarea routing), but many of the smaller internal routing operations, such
as recalculating the database –re-running the SPF algorithm, are
restricted within an area
OSPF Uses “Areas”
Changes in one area are
generally not propagated
(spread) to another
Route summarization is
extensively used in multiarea OSPF
OSPF Router Types
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.
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
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
OSPF and IS – IS Similarities
• Classless
• Link-state databases an Dijkstra’s 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
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.
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
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
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
IS – IS is Hierarchical
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
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
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 used?
• load
• Reliability
These values are used when the administrator
manually enters them
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)
EIGRP Tables
The following three tables are maintained by
EIGRP:
• Neighbor table
• Topology table
• Routing table
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.
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.
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, connectionoriented properties from TCP.
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
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.
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 governmentowned.
• 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.
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 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 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.
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.
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
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.
• 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
•
•
•
•
•
•
•
•
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
•
•
•
•
•
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
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
Making a Call
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
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 qualityand
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
Upcoming Deadlines
• Assignment 1-4-3 Data Center Design Project
Phase 3: Data Center Network Design is due
July 12
• Assignement 10-1 Concept Questions 7 is due
July 5
• Assignment 11-1 Concept Question 8 is due
July 12