Introduction To Packet Voice Technologies

Introduction to Packet
Voice Technologies
Presentation_ID
2008 Cisco Systems, Inc. All rights reserved.
Cisco Public
Communications at the beginning
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Evolution of the telephone
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Basic Components of a Telephony Network
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Basic Components of a Telephony Network
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Basis Components of a Telephony Network

PSTN: Public switched telephone network
Analog telephone: Able to connect directly to the PSTN
and is the most common device on the PSTN. Converts
audio into electrical signals.
Digital telephone: Typically connects to a PBX system.
Converts audio into binary 1s and 0s, which allows more
efficient communication than analog.
Private switch: Allows a business to operate a miniature
PSTN inside its company. This provides efficiency and
cost savings because each phone in the company does
not require a direct connection to the CO switch.
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Basis Components of a Telephony Network

Central office (CO) switch: Provides services to the
devices on the local loop. These services include
signaling, digit collection, call routing, setup, and teardown.
Trunk: Provides a connection between switches. These
switches could be CO or private.
Local loop: The link between the customer premises
(such as a home or business) and the telecommunications
service provider
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Local Loops, Trunks, and Interswitch

Communication
The communication path between several central office
switches is known as a trunk.
Meshed Network Versus Hierarchical Network

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Circuit-Switching Hierarchy
Switches are currently deployed in hierarchies. End office

switches (or CO switches) interconnect through trunks to
tandem switches (also referred to as Class 4 switches).
Higher-layer tandem switches connect local tandem switches
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PSTN Services and Applications
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PSTN Services and Applications
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Central Office Switches
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What is a PBX?
PBX: Private Branch Exchange
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Components of a PBX
Control complex: Provides the intelligence behind the PBX
system; all call setup,routing, and management functions
are contained in the control complex.
Line cards: Provide the connection between telephone
handsets and the PBX system.
Trunk cards: Provide connections from the PBX system to
the PSTN or other PBX systems.
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Basic Call Setup
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Loop Start Signaling (FXS)

On-hook,
open loop
Station
PBX or Central Office

Loop
(Local or Station)
BELL
Switch
Current
sense
48v
48v
48v
Off-hook,
close loop
DC Current
Switch
BELL
Ring on-hook
Ans off-hook
AC
BELL !!
Ringing
BELL
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Basic Call Progress: Idle

On-Hook
Open
Circuit
On-Hook
Open
Circuit
Telephone
Switch
Local
Loop
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Local
Loop
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Basic Call Progress: Dialing

Off-Hook
Closed
Circuit
On-Hook
Open
Circuit
dialtone
Telephone
Switch
DC Current
48v
Local
Loop
Dialed Digits
Pulses or Tones
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Basic Call Progress: Switching

Off-Hook
Closed
Circuit
Address
to
Port
Translation
On-Hook
Open
Circuit
?
Telephone
Switch
48v
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Local
Loop
21
Basic Call Progress: Ringing

Off-Hook
Closed
Circuit
90V AC
Ring Signal
Ring Back
Tone
On-Hook
Open
Circuit
RG
Telephone
Switch
Local
Loop
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Local
Loop
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Basic Call Progress: Talking

Off-Hook
Closed
Circuit
Off-Hook
Closed
Circuit
Voice Energy
DC Current
Local
Loop
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X
RG
Telephone
Switch
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Voice Energy
DC Current
Local
Loop
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Types of Local-Loop Signaling
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Supervisory Signaling
On-hook signal: When the phone is on-hook, the
connection between the tip and ring wires is broken and no
electrical signal passes between them.
Off-hook signal: When the phone is off-hook, the phone
connects the tip and ring wires, completing the circuit and
allowing electrical signal to pass.
Ringing: To cause an analog phone to ring, the phone

company sends an alternating current (AC) signal down
one of the wires, which the phone detects and generates a
ring signal.
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Informational Signaling
Dial tone
Busy
Ringback
Congestion
Reorder
Receiver off-hook
No such number
Confirmation
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Informational Signaling
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Address Signaling
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Address Signaling
Dual-tone multifrequency (DTMF): The buttons on a
telephone keypad use a pair of high and low electrical
frequencies (thus dual-tone) to generate a signal each
time a caller presses a digit. DTMF is the predominant
signal type used in the United States.
Pulse: The rotary-dial wheel of a phone connects and
disconnects the local loop circuit as it rotates around to
signal specific digits.
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Dual Tone Multifrequency
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Pulse Dialing
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Digital vs Analog Connections
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Sending Multiple Calls over a Single Line

The original problems of analog connections:
The signal degrades over long distances.
You cant send multiple calls over a single line (resulting in

massive cabling requirements).
Digitizing voice solves the first problem because you can easily
transmit a numeric value any distance a cable can run without
any degradation or line noise.
Multiplexing solves the second problem.
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FDM: Frequency-Division Multiplexing
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Hierarchy FDM
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TDM: Time-Division Multiplexing
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TDM: Time-Division Multiplexing
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Hierarchy TDM
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Drawbacks to the PSTN

Data has overtaken voice as the primary traffic on
networks.
The PSTN cannot create and deploy features quickly

enough.
Data/Voice/Video (D/V/V) cannot converge on the PSTN
as currently built.
The architecture built for voice is not flexible enough to
carry data.
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VoIP Overview
VoIP is the family of technologies that allows IP networks to

be used for voice applications, such as telephony, voice
instant messaging, and teleconferencing.
VoIP defines a way to carry voice calls over an IP network,
including the digitization and packetization of the voice
streams.
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VoIP Overview
VoIP services convert your voice into a digital signal that
travels over an IP-based network.
If you are calling a traditional phone number, the signal is
converted to a traditional telephone signal before it reaches its
destination.
VoIP allows you to make a call directly from a computer, a VoIP
phone, or a traditional analog phone connected to a special
adapter. In addition, wireless hot spots in locations such as
airports, parks, and cafes that allow you to connect to the
Internet might enable you to use VoIP services.
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VoIP Advantages
Reduced cost of communicating: Instead of relying on
expensive tie lines or toll charges to communicate between
offices, VoIP allows you to forward calls over WAN
connections.
Reduced cost of cabling: VoIP deployments typically cut
cabling costs in half by running a single Ethernet connection
instead of both voice and data cables. (This cost savings is
most realized in newly constructed offices.)
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VoIP Advantages
Take your phone with you: Cost estimates for moves, adds,
and changes (MAC) to a traditional PBX system range from
$55 to $295 per MAC. With VoIP phone systems, this cost is
virtually eliminated.
In addition, IP phones are becoming increasingly plug-and-play

within the local offices, allowing moves with little to no
reconfiguration of the voice network. In addition, when
combined with a VPN configuration, users can take IP phones
home with them and retain their work extension.
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VoIP Advantages
IP SoftPhones: SoftPhones represent an ideal example of the
possibilities when combining voice and data networks. Users can
now plug a headset into their laptop or desktop and allow it to act
as their phone. SoftPhones are becoming increasingly more
integrated with other applications such as e-mail contact lists,
instant messenger, and video telephony.
Unified e-mail, voicemail, fax: All messaging can be sent to a
users e-mail inbox. This allows users to get all messages in one
place and easily reply, forward, or archive messages.
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VoIP Advantages
Increased productivity: VoIP extensions can forward to ring
multiple devices before forwarding to voicemail.
Feature-rich communications: Because voice, data, and video
networks have combined, users can initiate phone calls that
communicate with or invoke other applications from the voice or
data network to add additional benefits to a VoIP call. For
example, calls flowing into a call center can automatically pull up
customer records based on caller ID information or trigger a
video stream for one or more of the callers.
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VoIP Advantages
Open, compatible standards: In the same way that you can
network Apple, Dell, and IBM PCs together, you can now connect
devices from different telephony vendors together. Although this
benefit has yet to be fully realized, this will allow businesses to
choose the best equipment for their network, regardless of the
manufacturer.
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VoIP Advantages
Flexibility: The sophisticated functionality of IP networks allows
organizations to be flexible in the types of applications and
services they provide to their customers and users. Service
providers can easily segment customers. This helps them to
provide different applications, custom services, and rates
depending on traffic volume needs and other customer-specific
factors.
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VoIP Advantages
Advanced call routing: When multiple paths exist to connect a
call to its destination, some of these paths might be preferred
over others based on cost, distance, quality, partner handoffs,
traffic load, or various other considerations.
Least-cost routing and time-of-day routing are two examples of
advanced call routing that can be implemented to determine the
best possible route for each call.
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VoIP Advantages
Long-distance toll bypass: Long-distance toll bypass is an
attractive solution for organizations that place a significant
number of calls between sites that are charged traditional longdistance fees. In this case, it might be more cost effective to use
VoIP to place those calls across an IP network. If the IP WAN
becomes congested, calls can overflow into the PSTN, ensuring
that no degradation occurs in voice quality.
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Circuit Switching
Dedicated communication path between two stations

(circuit)
Send on receives the whole data for communication through
this circuit
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Circuit Switching phases

Circuit establishment
Data transfer
Circuit disconnect
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Packet Switching
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Packet Switching Phases
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The whole data is split into small packets

The packet is transmitted into the network
The packet travels to the destination
The received packets are assembled back to the
whole data
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Packet Telephony vs. Circuit Switched
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Circuit Switching vs Packet Switching
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Call Control
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Call Control
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Distributed Call Control
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Centralized Call Control
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Packet Telephony Components
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Gatekeeper: Provides Call Admission Control (CAC),
bandwidth control and management, and address
translation.
Gateway: Provides translation between VoIP and nonVoIP networks, such as the PSTN.
Multipoint Control Unit (MCU): Provides real-time
connectivity for participants in multiple locations to attend
the same videoconference or meeting.
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Call agent: Provides call control for IP phones, CAC,
bandwidth control and management, and address
translation. Unlike a gatekeeper, which in a Cisco
environment typically runs on a router, a call agent typically
runs on a server platform.
Application servers: Provide services such as voice mail,
unified messaging, and Cisco Communications Manager
Attendant Console.
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Real-Time vs. Best-Effort Traffic
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Physical Connectivity Options
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Connections to the PSTN
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Trunks
Trunks are used to interconnect gateways or PBX systems
to other gateways, PBX systems, or the PSTN.
A trunk is a single physical or logical interface that contains
several physical interfaces and connects to a single
destination. This could be a single FXO port that provides a
single line connection between a Cisco gateway and a FXS
port of small PBX system, a POTS device, or several T1
interfaces with 24 lines each in a Cisco gateway providing
PSTN lines to several hundred subscribers.
Trunk ports can be analog or digital and use a variety of
signaling protocols
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Foreign Exchange Station (FXS) Interface
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Foreign Exchange Office (FXO) Interface
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E&M Interface
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E&M Interface
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Analog Trunks
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T1 Interface
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E1 Interface
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BRI Interface
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Connectivity Options
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Campus LAN Environment
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Centralized Call Processing Environment
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Distributed Call Processing Environment
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Service Provider Environment
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Analog-to-Digital Voice Encoding
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Facts about Sounds

The average human ear is able to hear frequencies
from 2020,000 Hz.
Human speech uses frequencies from 2009000 Hz.
Telephone channels typically transmit frequencies from

3003400 Hz.
The Nyquist theorem is able to reproduce frequencies
from 3004000 Hz.
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Digitizing Analog Signals
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1.-
Sample the analog signal regularly.
2.-
Quantize the sample.
3.-
Encode the value into a binary expression.
4.-
Compress the samples to reduce bandwidth,

optional step.
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Sampling Analog Voice
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Nyquist Theorem
If you sample at twice the highest frequency on a voice line,

you achieve good-quality voice transmission.
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Quantization
The process of converting the analog wave into digital,

numeric values is known as quantization.
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Quantization Techniques
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Converting Digital to Analog
1.-
Decompress the samples, if compressed.
2.-
Decode
the
samples
into
voltage
amplitudes, rebuilding the PAM signal.

3.-
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Filter the signal to remove any noise.
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Quantization Error
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Codecs
Codecs provide the coding and decoding translation
between analog and digital facilities. Each codec type
defines the method of voice coding and the compression
mechanism that is used to convert the voice stream.
The PSTN uses TDM to carry each voice call. Each voice
channel reserves 64 kbps of bandwidth and uses the G.711
codec to convert an analog voice wave to a 64-kbps
digitized voice stream.
In VoIP design, codecs might compress voice beyond the
64-kbps voice stream to allow more efficient use of network
resources.
The most widely used codec in the WAN environment is
G.729, which compresses the voice stream to 8 kbps.
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Compression Bandwidth Requirements
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Mean Opinion Score
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Mean Opinion Score
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Perceptual Speech Quality Measurement

Although MOS scoring is a subjective method of
determining voice quality, it is not the only method for doing
so. The ITU-T put forth recommendation P.861, which
covers ways you can objectively determine voice quality
using Perceptual Speech Quality Measurement (PSQM).
PSQM has many drawbacks when used with voice codecs
(vocoders). One drawback is that what the "machine" or
PSQM hears is not what the human ear perceives. In
layman's terms, a person can trick the human ear into
perceiving a higher-quality voice, but a computer cannot be
tricked. Also, PSQM was developed to "hear" impairments
caused by compression and decompression and not packet
loss or jitter.
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Perceptual Speech Quality Measurement
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Introduction To Packet Voice Technologies

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Introduction to Packet

2008 Cisco Systems, Inc. All rights reserved.

Communications at the beginning

2008 Cisco Systems, Inc. All rights reserved.

Evolution of the telephone

2008 Cisco Systems, Inc. All rights reserved.

2008 Cisco Systems, Inc. All rights reserved.

2008 Cisco Systems, Inc. All rights reserved.

Basic Components of a Telephony Network

2008 Cisco Systems, Inc. All rights reserved.

Basic Components of a Telephony Network

2008 Cisco Systems, Inc. All rights reserved.

Basis Components of a Telephony Network

2008 Cisco Systems, Inc. All rights reserved.

Basis Components of a Telephony Network

2008 Cisco Systems, Inc. All rights reserved.

Local Loops, Trunks, and Interswitch

Meshed Network Versus Hierarchical Network

2008 Cisco Systems, Inc. All rights reserved.

Switches are currently deployed in hierarchies. End office

2008 Cisco Systems, Inc. All rights reserved.

PSTN Services and Applications

2008 Cisco Systems, Inc. All rights reserved.

PSTN Services and Applications

2008 Cisco Systems, Inc. All rights reserved.

Central Office Switches

2008 Cisco Systems, Inc. All rights reserved.

PBX: Private Branch Exchange

2008 Cisco Systems, Inc. All rights reserved.

2008 Cisco Systems, Inc. All rights reserved.

Basic Call Setup

2008 Cisco Systems, Inc. All rights reserved.

Loop Start Signaling (FXS)

PBX or Central Office

2008 Cisco Systems, Inc. All rights reserved.

Basic Call Progress: Idle

2008 Cisco Systems, Inc. All rights reserved.

Basic Call Progress: Dialing

2008 Cisco Systems, Inc. All rights reserved.

Basic Call Progress: Switching

2008 Cisco Systems, Inc. All rights reserved.

Basic Call Progress: Ringing

2008 Cisco Systems, Inc. All rights reserved.

Basic Call Progress: Talking

2008 Cisco Systems, Inc. All rights reserved.

Types of Local-Loop Signaling

2008 Cisco Systems, Inc. All rights reserved.

Ringing: To cause an analog phone to ring, the phone

2008 Cisco Systems, Inc. All rights reserved.

2008 Cisco Systems, Inc. All rights reserved.

2008 Cisco Systems, Inc. All rights reserved.

2008 Cisco Systems, Inc. All rights reserved.

2008 Cisco Systems, Inc. All rights reserved.

Dual Tone Multifrequency

2008 Cisco Systems, Inc. All rights reserved.

2008 Cisco Systems, Inc. All rights reserved.

Digital vs Analog Connections

2008 Cisco Systems, Inc. All rights reserved.

Sending Multiple Calls over a Single Line

You cant send multiple calls over a single line (resulting in

Multiplexing solves the second problem.

2008 Cisco Systems, Inc. All rights reserved.

FDM: Frequency-Division Multiplexing