01.- introduction to packet voice technologies(1)

8/10/2019 01.- Introduction to Packet Voice Technologies(1)

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2008 C isco Sys tems, Inc. All r ights reserved. C isco Publi cPresentation_ID 11

Introduction to PacketVoice Technologies


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2008 C isco Sys tems, Inc. All r ights reserved. C isco Publi cULg VoIP 2

Communications at the beginning


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Evolution of the telephone


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5/96 2008 C isco Sys tems, Inc. All r ights reserved. C isco Publi cULg VoIP 5



Basic Components of a Telephony Network



Basis Components of a Telephony Network

PSTN: Public switched telephone network

Analog telephone: Able to connect directly to the PSTNand is the most common device on the PSTN. Convertsaudio into electrical signals.

Digital telephone: Typically connects to a PBX system.Converts audio into binary 1s and 0s, which allows moreefficient communication than analog.

Private switch:Allows a business to operate a miniaturePSTN inside its company. This provides efficiency andcost savings because each phone in the company doesnot require a direct connection to the CO switch.



Basis Components of a Telephony Network

Central office (CO) switch: Provides services to thedevices on the local loop. These services includesignaling, digit collection, call routing, setup, and teardown.

Trunk: Provides a connection between switches. Theseswitches could be CO or private.

Local loop: The link between the customer premises(such as a home or business) and the telecommunicationsservice provider



Local Loops, Trunks, and InterswitchCommunication

The communication path between several central officeswitches is known as a trunk.

Meshed Network Versus Hierarchical Network



Circuit-Switching Hierarchy

Switches are currently deployed in hierarchies. End office

switches (or CO switches) interconnect through trunks totandem switches (also referred to as Class 4 switches).Higher-layer tandem switches connect local tandem switches



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 PBXsystem; all call setup,routing, and management functionsare contained in the control complex.

Line cards: Provide the connection between telephonehandsets and the PBX system.

Trunk cards: Provide connections from the PBX system tothe PSTN or other PBX systems.


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Basic Call Setup


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Loop(Local or Station) +

48v

Station PBX or Central Office

Switch

Loop Start Signaling (FXS)

T

R

On-hook,

open loop

BELL

+

DC Current Switch48v

Off-hook,

close loop

BELL

+

ACRinging Switch

BELL ! !

48vBELL

Ring on-hook

Ans off-hook

Current

sense


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On-Hook

Open

Circuit

Basic Call Progress: Idle

Telephone

Switch

Local

Loop48vLocal

Loop

On-Hook

Open

Circuit


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DC Current

Dialed Digits

Pulses or Tones

Off-Hook

Closed

Circuit

Basic Call Progress: Dialing

Telephone

Switch

Local

Loop48v

dialtone

On-Hook

Open

Circuit


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Local

Loop

Off-Hook

Closed

Circuit

Basic Call Progress: Switching

Address

to

Port

Translation

Telephone

Switch

48v

?

On-Hook

Open

Circuit


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Local

Loop

Local

Loop

Basic Call Progress: Ringing

Ring Back

Tone

Off-Hook

Closed

Circuit

Telephone

Switch

On-Hook

Open

Circuit90V AC

Ring Signal

48v

RG


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Voice Energy

DC Current

Local

Loop

Local

Loop

Off-Hook

Closed

Circuit

Basic Call Progress: Talking

Telephone

Switch

Voice Energy

DC Current

Off-Hook

ClosedCircuit

48v

RG

X


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Types of Local-Loop Signaling


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Supervisory Signaling

On-hook signal: When the phone is on-hook, theconnection between the tip and ring wires is broken and noelectrical signal passes between them.

Off-hook signal: When the phone is off-hook, the phone

connects the tip and ring wires, completing the circuit andallowing electrical signal to pass.

Ringing: To cause an analog phone to ring, the phonecompany sends an alternating current (AC) signal down

one of the wires, which the phone detects and generates aring 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 atelephone keypad use a pair of high and low electricalfrequencies (thus dual-tone) to generate a signal eachtime a caller presses a digit. DTMF is the predominantsignal type used in the United States.

Pulse: The rotary-dial wheel of a phone connects anddisconnects the local loop circuit as it rotates around tosignal 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 easilytransmit a numeric value any distance a cable can run withoutany 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

Hierarchy TDM


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Hierarchy TDM


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Drawbacks to the PSTN

Data has overtaken voice as the primary traffic onnetworks.

The PSTN cannot create and deploy features quicklyenough.

Data/Voice/Video (D/V/V) cannot converge on the PSTNas 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 tobe used for voice applications, such as telephony, voiceinstant messaging, and teleconferencing.

VoIP defines a way to carry voice calls over an IP network,including the digitization and packetization of the voicestreams.


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VoIP Overview

VoIP services convert your voice into a digital signal thattravels over an IP-based network.

If you are calling a traditional phone number, the signal isconverted to a traditional telephone signal before it reaches its

destination.

VoIP allows you to make a call directly from a computer, a VoIPphone, or a traditional analog phone connected to a specialadapter. In addition, wireless hot spots in locations such as

airports, parks, and cafes that allow you to connect to theInternet might enable you to use VoIP services.


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VoIP Advantages

Reduced cost of communicating: Instead of relying onexpensive tie lines or toll charges to communicate betweenoffices, VoIP allows you to forward calls over WANconnections.

Reduced cost of cabling: VoIP deployments typically cutcabling costs in half by running a single Ethernet connectioninstead of both voice and data cables. (This cost savings ismost 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 isvirtually eliminated.

In addition, IP phones are becoming increasingly plug-and-playwithin the local offices, allowing moves with little to noreconfiguration of the voice network. In addition, whencombined with a VPN configuration, users can take IP phoneshome with them and retain their work extension.


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VoIP Advantages

IP SoftPhones: SoftPhones represent an ideal example of thepossibilities when combining voice and data networks. Users cannow plug a headset into their laptop or desktop and allow it to actas their phone. SoftPhones are becoming increasingly moreintegrated 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 ausers e-mail inbox. This allows users to get all messages in oneplace and easily reply, forward, or archive messages.


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VoIP Advantages

Increased productivity: VoIP extensions can forward to ringmultiple devices before forwarding to voicemail.

Feature-rich communications: Because voice, data, and videonetworks have combined, users can initiate phone calls that

communicate with or invoke other applications from the voice ordata network to add additional benefits to a VoIP call. For example, calls flowing into a call center can automatically pull upcustomer records based on caller ID information or trigger avideo stream for one or more of the callers.


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VoIP Advantages

Open, compatible standards: In the same way that you cannetwork Apple, Dell, and IBM PCs together, you can now connectdevices from different telephony vendors together. Although thisbenefit has yet to be fully realized, this will allow businesses tochoose the best equipment for their network, regardless of themanufacturer.


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VoIP Advantages

Flexibility: The sophisticated functionality of IP networks allowsorganizations to be flexible in the types of applications andservices they provide to their customers and users. Serviceproviders can easily segment customers. This helps them toprovide different applications, custom services, and ratesdepending on traffic volume needs and other customer-specific

factors.


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VoIP Advantages

Advanced call routing: When multiple paths exist to connect acall to its destination, some of these paths might be preferredover 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 ofadvanced 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 anattractive solution for organizations that place a significantnumber of calls between sites that are charged traditional long-distance fees. In this case, it might be more cost effective to useVoIP to place those calls across an IP network. If the IP WANbecomes congested, calls can overflow into the PSTN, ensuring

that no degradation occurs in voice quality.

Circuit Switching


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Circuit Switching

Dedicated communication path between two stations(circuit)

Send on receives the whole data for communication throughthis circuit


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Circuit Switching phases

Circuit establishment Data transfer Circuit disconnect


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Packet Switching

S


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Packet Switching Phases

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

P k t T l h Ci it S it h d


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Packet Telephony vs. Circuit Switched

Ci it S it hi P k t S it hi


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Circuit Switching vs Packet Switching


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Call Control

Call Control


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Centralized Call Control


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Centralized Call Control

Packet Telephony Components


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Gatekeeper: Provides Call Admission Control (CAC),bandwidth control and management, and addresstranslation.

Gateway: Provides translation between VoIP and non-VoIP networks, such as the PSTN.

Multipoint Control Unit (MCU): Provides real-timeconnectivity for participants in multiple locations to attendthe same videoconference or meeting.



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Call agent: Provides call control for IP phones, CAC,bandwidth control and management, and addresstranslation. Unlike a gatekeeper, which in a Ciscoenvironment typically runs on a router, a call agent typicallyruns on a server platform.

Application servers: Provide services such as voice mail,unified messaging, and Cisco Communications Manager

Attendant Console.

Real-Time vs Best-Effort Traffic


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Real-Time vs. Best-Effort Traffic


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Physical Connectivity Options

Connections to the PSTN


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Connections to the PSTN

Trunks


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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 containsseveral physical interfaces and connects to a singledestination. This could be a single FXO port that provides a

single line connection between a Cisco gateway and a FXSport of small PBX system, a POTS device, or several T1interfaces with 24 lines each in a Cisco gateway providingPSTN lines to several hundred subscribers.

Trunk ports can be analog or digital and use a variety of signaling protocols

Foreign Exchange Station (FXS) Interface


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Foreign Exchange Station (FXS) Interface

Foreign Exchange Office (FXO) Interface


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Foreign Exchange Office (FXO) Interface

E&M Interface


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E&M Interface


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Analog Trunks


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g

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

Facts about Sounds


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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 frequenciesfrom 3004000 Hz.

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.

Sampling Analog Voice


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Nyquist Theorem


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If you sample at twice the highest frequency on a voice line,you achieve good-quality voice transmission.



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Quantization


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The process of converting the analog wave into digital,numeric values is known as quantization.

Quantization Techniques


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Converting Digital to Analog


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1.- Decompress the samples, if compressed.

2.- Decode the samples into voltage

amplitudes, rebuilding the PAM signal.

3.- Filter the signal to remove any noise.

Quantization Error


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CodecsCodecs provide the coding and decoding translation


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Codecs provide the coding and decoding translationbetween analog and digital facilities. Each codec type

defines the method of voice coding and the compressionmechanism that is used to convert the voice stream.

The PSTN uses TDM to carry each voice call. Each voicechannel reserves 64 kbps of bandwidth and uses the G.711

codec to convert an analog voice wave to a 64-kbpsdigitized voice stream.

In VoIP design, codecs might compress voice beyond the64-kbps voice stream to allow more efficient use of network

resources.The most widely used codec in the WAN environment isG.729, which compresses the voice stream to 8 kbps.

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


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Although MOS scoring is a subjective method ofdetermining voice quality, it is not the only method for doingso. The ITU-T put forth recommendation P.861, whichcovers ways you can objectively determine voice qualityusing Perceptual Speech Quality Measurement (PSQM).

PSQM has many drawbacks when used with voice codecs(vocoders). One drawback is that what the "machine" orPSQM hears is not what the human ear perceives. Inlayman's terms, a person can trick the human ear intoperceiving a higher-quality voice, but a computer cannot be

tricked. Also, PSQM was developed to "hear" impairmentscaused by compression and decompression and not packetloss or jitter.

Perceptual Speech Quality Measurement


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01.- introduction to packet voice technologies(1)

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