The document discusses the evolution of telephony from analog to digital systems and the development of mobile computing through telephone networks. It covers topics like manual and automated telephone exchanges, pulse code modulation (PCM) for digital audio transmission, public switched telephone systems (PSTN), multiple access techniques like FDMA, TDMA, CDMA and SDMA. It also discusses interactive voice response (IVR) systems, voiceXML for internet-based voice applications and elements of building voice applications like prompts, grammars and forms.

1. MOBILE COMPUTING
THROUGH
TELEPHONY
Prakash Patil
B.V.B College of Engg, & Tech, Hubli

2. 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.

3. 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.

4. 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.

5. 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

6. 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

7. 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.

8. 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.

9. 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.

10. 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.

11. 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

12. Multiple Access Procedures

13. 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)

14. 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

15. 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’

16. 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.

17. 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

18. 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

19. 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

20. 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

21. IVR Architecture
 Call received by the voice card within IVR
 Voice card answer the call
 Establish the connection between the caller &
IVR application

22. 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

23. IVR Architecture
 Alphabets mapping on the telephone
keyboard

24. 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)

25. 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

26. 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

27. 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.

28. 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.

29. 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

30. 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.

31. 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

32. 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

33. 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

34. 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

35. 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.

36. Voice APIs
 Dialogic provides different APIs.

37. 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.

38. Call Flow for Theater Ticket Booking
Application

39. 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.

40. 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.

41. 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

42. Architectural Model
 A document server (Web server)
 Applications run on VoiceXML interpreter context
 Server delivers VoiceXML documents which are
processed by the VoiceXML interpreter.

43. Architectural Model
 VoiceXML interpreter Context:
 Responsible for detecting an incoming call
 Acquiring the initial voice XML document
 Answering the call

44. 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.

45. 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.

46. 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.

47. 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.

48. 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.

49. Essential Elements of Voice XML
Documents
 First line contains <?xml> element
 Second line contains <vxml> element

50. 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.

51. 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

52. Grammar-inline Grammars
 Inline Grammars: defined in the VoiceXML
code
 Example:
 Inline Grammars: defined in the VoiceXML
code
 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.

53. 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:

54. 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.

55. Form Example
 In VoiceXML forms are define using <form> element and
fields within the form element using the <field> element.

56. 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.


57. 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.

58. VoiceXML Elements

59. VoiceXML Elements

60. VoiceXML Elements

61. VoiceXML Elements

62. 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.

63. 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.

64. 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

65. Questions and
Assignments