Options in voice communication

By H Raghavan
CCCF
7 Aug 2015
Outline of the discussion
Information on present voice
infrastructure
Facts about existing system
Introduction to emerging technology
Comparison of new v/s traditional system
Mobile option
Work done by CCCF on new technology
Demo
 Information on present voice
infrastructure
Alcatel Omni PCX Enterprise 4400 system
1600 extensions: 1000 analog, 600 digital
Voicemail facility to serve 1000 extensions
~300 ADSL ports for internet connection
to colony
 Facts on present voice infrastructure
Purchased and installed 10 years back
System is robust and no major breakdown
since installation
ADSL is integrated with the system
A team of three manage the service
Support by service provider is good
1300 extensions in use currently
Introduction to emerging technology
VoIP
Abstract
What is VoIP
How VoIP works
VoIP architecture
VoIP components
Abstract
VoIP is a technology that enables users to
make and receive calls over a data circuit
(internet) instead of circuit switched
network (Public Switched Telephone
Network, PSTN)
 What is VoIP
VoIP is also called internet telephony or IP
telephony.
VoIP is a service / software application
which digitizes the analog audio signals to
IP packets and sends it over an IP network.
VoIP service allow users to make / receive
calls to / from any VoIP device / PSTN /
mobile number.
 How VoIP works

The main components of VoIP (Hardware

and software) are:
VoIP application software
Protocol
Hard / Soft phones
Media gateway
Codec
 VoIP application software
Open source and proprietary VoIP application
software is available for Linux and Windows
Works as server to VoIP clients (IP Phone,
softphone etc.)
Functions as soft switch for call processing and
call establishment.
It acts as a gateway for external calls
It also maintains database of users and provide
flexibility to define dial plan
Billing feature and usage logs for each extension
Provides security features to secure the server
Gives all the features we get from a EPABX
Video session is possible if phone supports
 List of VoIP software
Open source
Asterix PBX
Free Switch
Free PBX
GNU Gatekeeper
Yate
Proprietary
3CX PBX
Avaya Application Server
Cisco Call Manager
Alcatel-Lucent
 List of VoIP clients
List of open source VoIP clients for desktop / laptop
o Ventrilo
o X-Lite
o MicroSIP
o Linphone
o Empathy

List of open source VoIP soft clients for mobile users

o RedPhone
o Sipdroid
o Yuilop
o iCall Mobile
 VoIP Protocols
SIP (Session Initiation Protocol) and H.323 signaling
protocols are available for VoIP deployment
Both are non-proprietary and handle registration,
signaling, establishing channels and call termination
SIP is widely used and similar to http
SIP supports audio and video
SIP works at application layer
SIP protocol used for signaling during call
establishment and tear down
RTP (Real Time Protocol) used for transporting
voice data
UDP (User Datagram Protocol) used for transport
of voice packets
SIP ids are easy to remember like e-mail ids
sip:user@voip.tifr.res.in:port
VoIP Network Model
SIP

RTP , RTCP

Transport Layer UDP , TCP

Network Layer IP

Datalink Layer

Physical Layer
 IP Hard and Soft phones
IP hard phones are available from Rs 3,000/= to
Rs 40,000/=
Some hard phone models comes with built-in
camera
High end phones allow users to check mails.
and SMS
Open source and proprietary soft phones are
available for different OS and mobile phones.
IP hard phones resemble traditional phones
No user training required to use the phone
IP hard phones
VoIP phone interface
 Media gateway
An active device for interfacing PSTN
lines to VoIP server.
Available in 8, 16 and 24 port
denominations
Converts analog signals to digital and vice
versa
H.323 protocol used between VoIP server
and media gateway for call handling
 Codec
Acronym for coder-decoder
Separate codec used for audio and video
One of the function is to compress the raw
audio / video to save bandwidth
Efficiency of codec determines the quality of
audio / video
Popular audio codecs G711, G722, G723,
G726, G729
Popular video codecs H261, H263, H264
PSTN: Public Switched Telephone Network
Schematic of a VoIP setup
Signal flow in VoIP
 How a VoIP phone is connected
Do not need additional data and power
point

POE enabled
VoIP

Single data point serve both PC & Phone

POE (Power Over Ethernet) provide
power for VoIP phone
Comparison of new v/s traditional system
Total expenses incurred till date for the EPABX
Cost of Alcatel EPABX
INR 5300000
Upgrade Cost of Software & Digital phones
INR 2275000
AMC for 10 Years @ Rs.4.0 Lacs/annum
INR 4000000
Cabling cost for building colony network
INR 2500000
Incidental Expense
INR 500000
Total expense over a period of 10 Yrs for 1600 INR 14575000
ports
Per port expense over a period of 10 Years INR 9100
 Comparison
VoIP (Open source) EPABX
Disruptive technology Fading technology
Stable and open source Gets locked to a proprietary
technology available technology
Single network for voice and data Separate voice network to be
maintained
Less initial investment, approx. Incur high initial investment
1/3rd the cost of EPABX and recurring maintenance
charges
No cost for expansion, cost is only Expansion costs towards
towards hard phone purchase purchase of prop. hardware
Upgrade at no cost Upgrade costs are substantial
Mobility is built in to the
technology Mobility not possible
 Comparison
VoIP (Open source)
After complete integration with
SIP trunking, overseas calls are
cheaper
Voice quality a step below
compared to EPABX level
Data network centric
Power outage can disturb the
service
Prone to network congestion
Needs less of power and space
Single team can maintain both the Separate team is required
service
EPABX
Overseas calls are expensive
Very high quality
Dedicated network
Service continues even under
long hours of power outage
Do not get affected by
external factors
Needs more power and space
Completely converged Network
Integrated TIFR voice communication system
Telecom norms in India for VoIP communication

Telecom regulator of India permits voice

communication over IP.
Framework of regulator permit communication
under the conditions
a) PC to PC (Both within as well as outside
India).
b) PC to Telephone (PC in India to Telephone
outside India)
c) IP based H.323/SIP Terminals in India to
similar terminals both in India and abroad,
employing IP addressing scheme of IANA
 Mobile option (Non-VoIP )
Mobile communication by availing CUG (Closed User Group)
feature. Calls within CUG is free.
This option needs a low investment and implementation is
quick
Considerable saving as there will be no AMC charges.
No voice network maintenance cost to TIFR, service will be
maintained by mobile operator
Yearly voice communication cost of TIFR Colaba campus is
around Rs 22 L to Rs 24 L. (Excluding official mobile charges)
Each user will be provided by a mobile phone instead of wired
phone as extension.
Guaranteed uptime and mobility feature built-in
To be proposed to CCCF committee after detail exploration
List price for each connection is Rs 99/= per month includes
free 200 mins talk time and 100 SMS
 Work done by CCCF on VoIP
Using open source VoIP software, platform is
built and ready for testing.
Tested VoIP features and call quality using
hard and soft phones.

Work to be done by CCCF on VoIP

Request to be submitted to CCCF committee
and seek its approval for beta testing.
Acquire skills for building redundant VoIP
system.
System to be tested on IPv6.
Acknowledgement
I thank my colleagues Sri Vijay Naik and
Sri Shisheer Teli for their contribution
 Live demonstration of VoIP features
Call a VoIP extension and check for voice
quality
Calling TIFR VoIP number from outside
Three party conference
Voice mail
Calling a user on his mobile having a VoIP
extension
Receiving call over TIFR wireless network
Multiple extensions ringing simultaneously
Video call over VoIP softphone
Thank you for your participation

H Raghavan
CCCF