MOBILE COMPUTING

THROUGH TELEPHONY
Prakash Patil
B.V.B College of Engg, & Tech, Hubli
Evolution of Telephony
 First telephone system developed by Alexandra Graham
Bell.
 Allowed two way voice communication between two
individuals in two locations on either side of a wire,
 Calling Party - A person who make a call
 Called party – A person who responds to the call
 During analog telephony, the purpose of
interconnecting two subscribers was to establish a
physical connection between their respective telephone
devices.
Evolution of Telephony
 In early days, each telephone was connected to a
central place – the exchange
 From exchange the operator would manually
connect to another subscriber.
 Billing information was maintained manually.
 Trunk Call - Call some one connected to other
exchange – The call would have to be set up with a
whole chain of operators, each one calling the next
and so on.
Evolution of Telephony
 1890 – development of the first automatic
telephone exchange- called “Stronger Switch” after
its originator Almon B Strowger.
 1982 – First version of automatic exchange was
installed to eliminate the human intervention.
 1912- Swedish engineer Gotthief patented an
automatic switching system based on a Grid –
Electromechanical and called crossbar exchange.
Evolution of Telephony
 1960 – Electronic Switching System (ESS) was
developed at AT & T labs.
 1962 – Carrier system was made digital.
 1976 – Bell labs developed 4ESS toll switch for the
long distance voice network (First Digital Circuit
Switch)
 1960-70s: Telephone exchanges controlled by
processors and software.
 1962 – Carrier System was made digital
Pulse Code Modulation (PCM)
 Used for Digital Modulation
 Audio Voice – 0-4 KHz.
 Measured amplitude converted to number
(Quantization process) i.e, represented by 8 bits.
 Snapshot of voice signal amplitude is taken at
1/8000th of Second (Double the frequency of
4KHz)
 1962- Bell lab introduced digital transmission
using PCM
Manual Exchange
 Manual Exchange – operator intelligence was the
control system
 An operator, alerted to an incoming call
 Listen to and remember desired number
 Finds the right way to connect the callers line to the line
being called
 Check if the desired line is free
 Makes the connection
 Note down the call details, time of call, duration of call,
calling number and called number.
Automated Exchange
 Indicates the progress of the call to the caller
 A series of distinct tones were generated by
machine called Ring Generator
 Dial Tone (DT) – Signal applied to the line after
calling party has lifted his handset.
 Busy Tone(BT) – Indicate route to called
subscriber is congested is OFF Hook.
 Ring Tone – Tone generated after circuit is
established between 02 parties.
PSTN- Public Switched Telephone System

 Normal Telephone System

 Also called – End Office or Local Access Tandem
 Local Exchange – used for the connection of
subscriber
 Transit Exchanges – Switch traffic between within
and different geographical areas.
 Local loop: A physical cable laid from the local
exchange to the telephone device at each subscriber
place – Called Last mile link.
 Multiple Access Procedures
 In PSTN- A separate wire is used to connect the
subscribers telephone with switch.
 Multiple users can have speech communication at the
same time without causing any interference to each
other.
 Unless we control the simultaneous access of radio
channel by users a collision can occur.
 Connection oriented communication the collision is
undesirable.
Multiple Access Procedures
 Every mobile subscriber must be assigned a
dedicated channel on demand.
 Achieved using different multiplexing techniques.
 FDMA- Frequency Division Multiple Access
 TDMA- Time Division Multiple Access
 CDMA- Code Division Multiple Access
 SDMA- Space Division Multiple Access
Multiple Access Procedures
FDMA- Frequency Division Multiple Access

 One of the most common Multiplexing Procedures.

 Available frequency band is divided into channels
of equal bandwidth
 Each communication carried on different
frequency.
 Used in all First generation analog mobile networks
like AMPS (Adv. Mobile Phone System) in USA
and TACS (Total Access Communication System in
UK)
TDMA- Time Division Multiple Access

 More expensive technique compared to FDMA

 Needs precise synchronization between transmitter
and receiver.
 Used in Digital Mobile Communication
 Whole frequency BW divided into sub-bands using
FDMA technique.
 TDMA is used in each of these sub-bands to offer
multiple access.
 GSM uses such a combination of FDMA and TDMA
CDMA- Code Division Multiple Access

 Broad band system

 Uses spread spectrum technique where each
subscriber uses whole system bandwidth
 All subscribers in a cell use the same frequency
band simultaneously.
 To separate the signal , each subscriber is assigned
an orthogonal code called ‘CHIP’
SDMA- Space Division Multiple Access

 Make use of space effectively

 Use different part of the space for multiplexing
 Used in radio transmission and more useful in satellite
communications to optimize the use of radio spectrum
by using the directional properties of antennas.
 In SDMA, antennas are highly directional, allowing
duplicate frequencies to be used at the same time for the
multiple surface zones on earth.
 Requires careful choice of zones for each transmitter and
also requires precise antenna alignment.
Mobile Computing through Telephone

 Accessing applications and services through voice

interface
 Referred as Computer Telephony Interface (CTI)
 Ex. Telephone Banking Application
 Input - a telephone Keyboard
 Output – Synthesized voice
Mobile Computing through Telephone

 Toll Free Service – Only one number is published

 Number is not attached to any exchange or specific
city
 Advantages
 User remember only one number
 Call same numbers from anywhere
 Numbers are generally toll free
 Need not worry about call charges
Mobile Computing through Telephone

 Interactive Voice Response (IVR)

 Voice Response Unit (VRU)
 Computer Telephony (CT)
 Computer Telephony Interface / Integration (CTI)
 IVR software can be hosted on WindowsNT, Linux
or any other computers with voice card
 One of the most popular card vendor is from
Intel /Dialogic
IVR Architecture
 IVR works as the gateway between a voice based
telephone system and computer system
 Multiple telephone lines are connected to the voice
card through telecom interface
IVR Architecture
 Call received by the voice card within IVR
 Voice card answer the call
 Establish the connection between the caller & IVR
application
IVR Architecture
 Switch can be either a PSTN exchange or local PBX
in the office
 Telephone KB has 12 keys (0,1,2,……...9, *,#)
 It is possible to enter alphabetic data through Tel. KB
IVR Architecture
 Alphabets mapping on the telephone keyboard
IVR Architecture
 It is possible to enter alphabetic data through
telephone KB by pressing a key in multiple
succession.
 Ex. DELHI entered as 3-3 (D), 3-3-3(E), 5-5-5-
5(L), 4-4-4-4(I)
 Key inputs are received by the voice card as
DTMP(Dual Tone Multi Frequency)
IVR Architecture
 Inputs generated through the combination of frequency
 Ex: User Press 2- 3Times (2-2-2 )
 Voice card will receive –
 697+1336Hz+ 697+1336Hz+ 697+1336Hz
 Looking at a time interval between the numbers the program
can decide whether the user entered 222 or B
IVR Architecture
 When application needs to send an output to the
user , the standard data is converted into voice
either through synthesizing voice files or through
Text to Speech (TTS) conversion software.
 IVR System assemble a series of prerecorded voice
prompts to generate equivalent sound response.
 TTS interface can be used to convert the text into
speech.
 Different TTS are available for different language.
Overview of the Voice Software
 Encompasses the processing and manipulation of an
audio signal in a Computer Telephony System (CT)
 Supports – Filtering, analyzing, recording, digitizing,
compressing, storing, replaying audio voice.
 Most of the voice cards come with industry standard
Peripheral Component Interface (PCI)
 PCI interface makes it possible to integrate these voice
products into Windows or Linux systems quite easily.
Overview of the Voice Software
 Encompasses the processing and manipulation of an
audio signal in a Computer Telephony System (CT)
 Supports – Filtering, analyzing, recording, digitizing,
compressing, storing, replaying audio voice.
 Most of the voice cards come with industry standard
Peripheral Component Interface (PCI)
 PCI interface makes it possible to integrate these voice
products into Windows or Linux systems quite easily.
Inside IVR
 Popular Voice card – D/41JCTLS from Dialogic for small office
 4 port analog converged communications voice, fax and software based speech
recognition board.
 Posses Dual Processor architecture
 Comprises Digital Signal Processor (DSP) and general purpose Microprocessor
 Provides four telephone line interface circuit for direct connection to analog
loop start lines RJ11 interface
 Voice Driver and API
 Dialogic Voice Driver APIs
 Many vendors around the world use Dialogic cards from Intel
in IVRS system.
 Driver in an IVRS system used to communicate and control
the voice hardware on IVRs System.
 Voice driver can make calls, answer calls, identify caller ID,
play and record sound from telephone line, detect DTMF
signals dialed by the caller.
 It can tear down the call, detect when the caller has hung up.
 It offers APIs to record the transaction details.
 Transaction information required for audit trail and cahrging.
IVR Programming
 Voice libraries provided by Dialogic to interface
with the voice driver.
 Single threaded and Multithreaded
 Libdxxmt.lib – the main voice library
 Libsrlmt.lib- Therstndard run time library
IVR Programming
 Use of Libraries
 Utilize all the voice board features of call management
 Write applications using single threaded Asynchronous
or multi threaded paradigm.
 Configure devices
 Handle the events that occur on the devices.
 Return device information
 Gather call transaction details
Single Threaded Asynchronous Programming Model

 Enable single program to control multiple voice

channels within a single thread.
 Allows development of complex applications
where multiple tasks must be coordinated
simultaneously
 Supports – Polled and callback event management
Multi Threaded synchronous Programming Model

 Application controls each channel from a separate

thread or process.
 Operating system can control individual device
threads to sleep
 When dialogic function is completed , the OS
wakes up the function’s thread so that processing
continues.
 Assign distinct applications to different channels
Voice APIs
 Dialogic provides different APIs.
 APIs are available for
 device management,
 configuration function,
 input output functions,
 play and record functions,
 tone detection functions,
 tone generation functions,
 call control functions etc.
Voice APIs
 Dialogic provides different APIs.
Developing IVR application
 User interface in IVRs application is called CALL
FLOW.
 Call Flow define – How call will be managed?
 Note down the precise prompts that are played as
output.
 Prompts are generally prerecorded by people with
professional voice.
Call Flow for Theater Ticket Booking Application
VoiceXML
 In Mobile Computing through telephony, IVR is
connected to the server through the Client / Server
Architecture.
 Today Internet (http) is used in addition to client /
server interface between IVR and Server in MC.
 http is used for voice portals.
 Increase the flexibility in MC architecture.
VoiceXML
 Voice Portal – A user use an Internet site through
voice or telephone interface.
 All these advanced features introduced VoiceXML.
 Recent IVRs are equipped with DSP (Digital
Signal Processing) & Capable of recognizing
voice.
 Output is synthesized voice through TTS.
VoiceXML- Voice eXtensible Markup Language

 XML based markup language for creating distributed

voice applications.
 Designed for creating audio dialogs.
 We can create web based voice applications that user
can access through telephone
 Features of VoiceXML
 Synthesized Speech
 Digitized Audio
 Recognition of Spoken English
 DTMF Key Input
Architectural Model
 A document server (Web server)
 Applications run on VoiceXML interpreter context
 Server delivers VoiceXML documents which are processed by
the VoiceXML interpreter.
Architectural Model
 VoiceXML interpreter Context:
 Responsible for detecting an incoming call
 Acquiring the initial voice XML document
 Answering the call
 How Voice XML Fits into Web Environment

 Visual GUI web browser renders and interprets http

requests to present information to the user in text,
multimedia, audio etc.
 The voice browser extends this paradigm.
 Voice server has been added to the web environment.
 How Voice XML Fits into Web Environment

 The Voice Server manages several Voice Browser

Sessions.
 Each Voice browser session includes one instance of
the Voice Browser, the speech recognitions engine, and
Text-to-Speech engine.
 How Voice XML Fits into Web Environment

 The voice browser presents the information to the

caller into audio using VoiceXML.
 When caller says something, the voice browser sends
HTTP request to the web server and information is
returned in Audio.- Called Voice Portal.
The Voice Browser
 Using voice browser, we can interact with web
server using our voice and a telephone.
 Voice browsers renders and interprets VoiceXML
document.
 We use voice and telephone (even phone keypad)
to access web information and services.
Dialogs
 A VoiceXML application defines a series of dialogs
between user and computer.
 Two types if dialogs that can be implemented in
VoiceXML
 Forms – Collects values for a set of fields
 Menus – presents user with choices or option and
transition to another dialog based on the choice.
Essential Elements of Voice XML Documents

 First line contains <?xml> element

 Second line contains <vxml> element
Prompts
 In Voice XML application, information is presented
to the user through Audio Prompts.
 Prompts can be prerecorded audio or synthesized
speech.
 Use prompt element in VoiceXML to generate TTS.
 Any text within the body of a <prompt> element is
spoken.
Grammar
 Each dialog has one or more speech and/or DTMF
grammars associated with it.
 In VoiceXML, <grammar> element is used to
define what the caller can say to the application at
any given time.
 Three different types of grammars supported.
 Inline
 External
 Built-in
Grammar-inline Grammars
 Inline Grammars: defined in the VoiceXML code
 Example:
 Words and phrases that caller is allowed to say is
defined within the body of the <grammar> element
 Each word is separated by “|” means OR.
Grammar-External Grammars
 External Grammar – are those specified outside of the
VoiceXML code.
 Document is in another file and referenced from within the
voiceXML code.
 <grammart> element is used to specify an external grammar.
 Example:
Form
 Form is one of the ways of developing a dialog
with the caller in VoiceXML.
 A form is, basically, a collection of one for more
fields that caller fills in by saying something.
 Forms are central to VoiceXML,
 A VoiceXML form is a process to present
information and gather input from the caller.
 In case of VoiceXMl, we can’t see the fields,
instead of typing into the field, we say something
to fill it in.
Form Example
 In VoiceXML forms are define using <form> element and
fields within the form element using the <field> element.
Events
 VoiceXML provides a form filling mechanism for
handling ‘normal’ user input.
 VoiceXML defines a mechanism for handling
events not covered by the form mechanism.
 Events are thrown by the platform under a variety
of circumstances, such as when the user does not
respond, does not respond intelligibly, request help
etc.

Links
 A link supports a mixed initiative.
 It specifies a grammar that is active whenever the
user is in scope of link.
 If the user input matches the link’s grammar
control transfer to the link destination URI
 A <link>can be used to throw an event to go to the
destination URI.
VoiceXML Elements
VoiceXML Elements
VoiceXML Elements
VoiceXML Elements
Telephony Application Programming Interface (TAPI)

 TAPI – Higher level framework for developing

voice services.
 SAPI- Speech Application Programming Interface
 SAPI and TAPI are two standards that can be used
uwhen developing voice telephony application.
 Developed jointly by Intel and Microsoft.
Telephony Application Programming Interface (TAPI)

 Advantages- Programmers can use different

telephone systems- PSTN, ISDN, and PBX(Private
Branch Exchange) without having to understand all
their details
 Use of API will save the programmers time and
pain of trying to program hardware directly
 Through TAPI and SAPI program can talk over
telephone or video phones or phone connected
resources.
Telephony Application Programming Interface (TAPI)

 Simple user interface to setup calls – calling some

one clicking the picture
 Simple GUI to setup a conference call
 See who you are talking
 Attach voice greeting with email
 Send and Receive faxes
Questions and Assignments
