Welcome

In this Presentation We will learn about the following

Topics
What is VoIP?
VoIP
Components
VoIP – Advantages and
Disadvantages
VoIP CODEC’s

Asterisk Based PBX

Communication System I R I S E T
 What is VOIP
 VOIP (Voice over Internet Protocol) is a technology that allows
telephone calls to be made over Internet.
 Packetization and Real-Time Transport of classic Telephone system
audio over an IP network.
 Allows 2-way voice transmission over broadband connection.
 Also called IP Telephony, Internet telephony, Broad-Band Telephony
etc
 Uses a specifically designed an ‘Open-Souce’ Protocol called ‘SIP’.
 Other Protocols like MGCP, skinny, Yate, IAX also support VoIP in
their respective PBX software.
I R I S E T
 Traditional Digital Telephony involves signalling, channel
setup, digitization of the analog voice signals, and encoding.
And are being transmitted over a circuit-switched network.

 In VOIP the digital information is packetized, and

transmission occurs as IP packets over a packet-switched
network and transportation of audio streams using special
media delivery protocols that encode audio and video with
audio codecs, and video codecs.

I R I S E T
 IP Exchanges, IP-Based Exchanges, IP Telephony servers,
Call Managers, Call Management server, IP-PBX’s are some
of the nomenclature given to a Telephony system which uses
partially or fully an Internet back bone and allows
transmission of voice over it by using a suitable protocol
supporting the call set, monitoring and releasing a call.

 We can say that a complete session is being established in

real time as it is in wired or circuit switching network.

 Because we are discussing about Telephony system, we must

feel the taste of telephony and its relevant instances on IP
network by using a IP phone. I R I S E T
Instances such as dial tone, ring back tone, busy tone or user busy
message and features like Camp On No Response, Camp On Busy, Call
Hold, Call Transfer or 3way conference etc.

Also we must also like to have a multiparty conference, voice mail,

Boss-Secretary and IVR facility in the IP Telephony system.

For getting the same taste of Telephony system, we must have a

Exchange Software to be installed on any computer or so called Server
and configured according to our needs.

I R I S E T
 IP Exchanges can of Hybrid type, upgraded to IP Platform.
 combination of TDM switch and IP switching including all relevant
Hardware such as Subscriber Gateways called FXS and FXO, Trunk
Gateways called as PRI30 or 30T gateways.
 They are attached as modules and other existing interfaces are also
kept in separate cabinets.
 There is a PBX Application software to run subscriber analog,
subscriber Digital, Soft phones and IP phones etc.
 In these exchanges the Core platform is based on TDM Technology.
I R I S E T
 VoIP - System
Components
 IP Exchange (Server) + OS
 VoIP Software
 IP Telephone ,IP Video Phone
 VoIP Gateway (Analog or Digital)
 Network Switches (with PoE)
 Cables and connectors
 Power Supply for various components

I R I S E T
 IP SERVER(EXCHANGE)
 This is a common server on which a VoIP software is
installed and configured to provide voice telephony.
 Some vendors sell propriety servers as well in the name of
features.
 Board has issued a policy guideline to use open standard
telephony.

I R I S E T
 VoIP SOFTWARE
 Open Source
– Asterisk, Freeswitch, SipXecs, yate etc.
 Propriety
– Cisco, Tadiran, Microsoft etc.

Open Source Telephony software :

 Open source software provides users the freedom of choice in
programming according to user requirements.

 It eliminates the vendor lock-in as well as promotes openness

and standardization such as (license for server software, license
for IP phone, license for gate ways , license for concurrent calls,
etc)
I R I S E T
 IP
TELEPHONE
S phone.
 Looks like the ordinary traditional
 Works like a modern mobile phone.
 Dial tone is local. Dial the number and then send the same to the
exchange.
 Requires LAN connectivity
 It can be powered with external DC power or can be PoE enabled
ethernet port
 The phone should be configured with account details and IP address.

I R I S E T
IP Phones

I R I S E T
 VoIP
GATEWAYS
 VoIP gateway is a device that converts telephony traffic
into a IP Packet for transmission over a data (IP) network
and vice-verse.
 Can be Analog or Digital.
 Analog VoIP Gateways are used to connect ordinary
Push-Button telephone to a VoIP system.
 Analog gateways – FXS, FXO

I R I S E T
GATE WAYS

I R I S E T
 GATEWAYS
 FXS is used for connecting the PBT (Analog) to the Gateway.
 FXO is used for connecting CO Line (dial tone from other
exchange) to the Gateway.
 Digital Gateways are used for connecting a PRI line to the VoIP
system.

I R I S E T
 Network Switches
 These are normal switches but support Power over Ethernet
(PoE) on each port.
 IEEE 802.3af-2003

 IEEE 802.3at-2009
30W of DC power/Power available – 25.5W

I R I S E T
 AUDIO CODECS in VOIP EXCHANGES

 Popular audio codecs

 μ-law and a-law versions of G.711, G.722,
 a popular open source voice codec known as
iLBC,
 a codec that only uses 8 kbit/s each way
called G.729

I R I S E T
 VOICE CODECS
 ADPCM: Adaptive Differential Pulse Code Modulation,
 LD-CELP: Low Delay Code Excited Linear Predictive
 CS-ACELP: Conjugates Structure Algebraic Code Excited
Linear Predictive,
 MP-MLQ: Multi Pulse, Multi Level Quantization
 MOS :Mean opinion score

I R I S E T
ADPCM
• Sends only the differential between the current and last sample
• Uses G.726 variants
• Allows encoding PCM data rates of 16 kb/s, 24 kb/s, or 32 kb/s per call
• Has lower quality than G.729
• Is not commonly used today
CS-ACELP
• Is based on the human vocal system
• Matches sounds to a codebook of possible sounds
• Uses G.729 variants
• Is the most common compression method used today
• Has a data rate of 8 kb/s per call
• Provides high quality

I R I S E T
 VOIP CODECS
 A codec transforms analog signals into digital format.
 Different codecs have different bandwidth requirements:
G.711: The G.711 codec uses the most bandwidth. It encodes each
of the 8000 samples that are taken each second into 8 bits, resulting
in a 64-kbps codec bandwidth.
G.722: The G.722 wideband codec splits the input signal into two
sub-bands and uses a modified version of adaptive differential
pulse code modulation (ADPCM) (including adaptive prediction)
for each band. The bandwidth of G.722 is 64, 56, or 48 kbps.
G.726: The G.726 ADPCM coding schemes uses less bandwidth.
These coding schemes encode each of the 8000 samples that are
taken each second using 4, 3, or 2 bits, resulting in bandwidths of
32, 24, or 16 kbps. I R I S E T
 VOIP CODECS

 G.728: The G.728 low-delay code excited linear prediction (LDCELP)

coding scheme compresses pulse code modulation (PCM) samples using
a codebook. Wave shapes of five samples are represented by a 10-bit
code word, which identifies the best matching pattern of the codebook.
Because of the compression of five samples (worth 40 bits in PCM) to
10 bits, the bandwidth of LDCELP is 16 kbps.
 G.729: The G.729 conjugate structure algebraic code excited linear
prediction (CS-ACELP) coding scheme also offers codebook-based
compression. Wave shapes of 10 bits are represented by a 10-bit code
word, reducing the bandwidth to 8 kbps.

I R I S E T
 ADVANTAGES of VOIP
 Portability :convenience of being able to make and receive calls from
any location using the same phone number VoIP takes the lead.
 Scalability: VoIP network is perfect for small and large business
communities can be expended at any time just by increasing the license
at server, and connecting additional voip phones to already established
LAN network.

I R I S E T
 Advantages of VOIP
 Flexibility: More flexible for implementation of the VoIP network. there
is no need for additional hardware for expansion (such as expansion of
shelves, cards, license, outdoor network, etc) since the only device
required for VoIP service performance is the VoIP phone system.
 COST: VoIP calls carried over the Internet are cheaper and can save a lot
of money especially for large enterprises that have to handle a huge
number of calls on daily basis
 Multi functionality: Call forwarding, call waiting, paging, group calls,
speed dialling and lots of other features deliver more enhanced call
processing opportunities that can bring to higher productivity.
I R I S E T
 Disadvantages of VOIP
 No service during a power outage :
 Reliability :your VOIP service will be affected by the quality and
reliability of your Network . Poor internet network and congestion can
result in garbled or distorted voice quality
 Security :Security is a main concern with VoIP, as it is with other Internet
technologies. The most prominent security issues over VoIP are identity
and service theft, viruses and malware, denial of service, spamming, call
tampering and phishing attacks
I R I S E T
 VOIP PROTOCOL STANDARDS
 ITU: International Telecommunication Union
H.323 -ITU recommends for “ Packet based Multimedia communication systems”.
- Most common VoIP protocol
- Distributed Architecture
 IETF: Internet Engineering Task Force
SIP: Session Initiation Protocol
-IETF RFC 2543 ,3265
-Distributed Architecture
 Real-time Transport Protocol – RTP
- A transport protocol for real-time application
- IETF RFC 1889 ,3550
- Provides transport for audio/media of VoIP communication
-Used by All of VoIP signalling protocols
I R I S E T
 VOIP PROTOCOL STANDARDS

MGCP Media Gateway Control Protocol

- IETF RFC 2075 ,3435
- Centralized Architecture for Multimedia applications such as
VoIP

H.248 Gateway Control Protocol

Collaboration between ITU & IETF referred to as IETF RFC
2885,3015 (MEGACO)

I R I S E T
Call Exchange (direct mode)

I R I S E T
 SIP
 Proposed Standard described in IETF RFC 2543
 Application-layer control protocol
 A signaling protocol for initiating, managing and terminating voice
and audio session across packet networks with one or more
participants .
 Text-based protocol with highly extensible
Session can be
 Call between two simple telephone
 Collaborative multi-media conference session, etc.

I R I S E T
 SIP
Functionality
 User location

 User availability
 User Capabilities
 Session setup
 Session Management
 Session termination

I R I S E T
 FUNCTIONALITIE
S
SIP serves 4 major functionalities
1 It allows to locate the user ( i.e translating user’s name to their
current network address)
2 Inviting the user for session -negotiation so that all of the participants
in a session can agree on the features to be supported among them
3 Delivering the session description - call management such as adding,
dropping or transferring participants.
4 Terminating the session.

I R I S E T
SIP ARCHITECTURE

I R I S E T
SIP ARCHITECTURE

I R I S E T
I R I S E T
 SIP ENTITIES

User Agent (UA)

 User agent client (UAC)
 User agent server (UAS)
Proxy server
 Stateless proxy server
 Statefull proxy server
Redirect server
Registrar server

I R I S E T
 USER
AGENT(UA)
 User Agent Client (UAC) –Application which originates SIP requests
 User Agent Server (UAS)
–Application which contacts user upon receiving SIP request, and–Returns user’s
response on his behalf
- Accepts, rejects or redirects
 User Agent (UA)
–Application which contains both UAC & UAS and exchange request/response
messages
 UA is a piece of software that can be placed in a computer or a laptop
Therefore, SIP can offer –Various telephony services,
e.g., ▪Internet phones-to-Internet phones
▪Internet phones-to-PSTN phones
▪PC phones-to-PC phones
I R I S E T
 SIP SERVERS
 proxy server: The Proxy Servers are application layer routers that
forward SIP request & responses
 Redirect server: A redirect server is a server that accepts a SIP
Requests & then return the location of another SIP user agent & server
where the user might be found.
 Registrar server: A registrar is a server that accepts REGISTER
requests.
 A registrar server is typically co-located with a proxy or redirect server
and offer location services

I R I S E T
 SIP Operation

1.invitation 2.invitation
4. OK 3. OK
USER 5.Acknowledge 6.Acknowledge USER
AGENT 1 AGEN
T2

7. Audio/Video data 7. Audio/Video data

I R I S E T
 SIP Protocols

SIP provides basic elements of telephony

SIP: Call setup and termination
RTCP: data stream management
DNS = Domain Name System
PPP = Point-to-Pont Protocol
RSTP = Real-time Streaming Protocol (controls video streams, like
a VCR)
RSVP = Resource Reservation Protocol
RTP = Real-Time Transfer Protocol
SDP = Session Description Protocol

I R I S E T
I R I S E T
 SIP
REQUESTS
INVITE –Request initiation of a session -Most common and
important
ACK –Confirm that a session has been initiated
BYE –Request termination of a session
OPTIONS –Query a host about its capabilities
CANCEL –Cancel a pending request
REGISTER –Inform a redirection server about the user’s
current location

I R I S E T
 SIP
RESPONSES
•1xx-Provisional (Informational) –Request received, continued to process request, e.g., 180 = Ringing

•2xx–Success –Action was successfully received, understood, and accepted, e.g., 200 = OK

•3xx –Redirection –Further action must be taken to complete the request, e.g., 305 = Use Proxy

•4xx-Client Error –The request contains bad syntax or cannot be fulfilled at the server,

e.g., 484 = Address Incomplete

•5xx -Server Error –The server failed to fulfill an apparently valid request,

e.g., 500 = Internal Server error

•6xx-Global Failure –The request is invalid on any server, e.g., 600 = Busy

I R I S E T
 Server and OS
requirements
Server for 100 subscribers for handling 50 concurrent calls

Server shall be suitable for 24X7 operations (Dell, IBM ,HP, etc)

Server should be installed in 1+1 redundancy for critical uses.

The processor should be intel dual core minimum1.8 GHz processor with
2 GB RAM,256 MB cache , RAID1 Dual hard disk 250GB minimum,
with dual Ethernet ports to enable redundant connection to LAN

OS shall be recent standard linux distribution.

I R I S E T
 Server and OS requirements

Server for 1000 to 1200 subscribers for handling 500 concurrent calls
Server shall be suitable for 24X7 operations (Dell, IBM ,HP, etc)
Server should be installed in 1+1 redundancy for critical uses .the second
server may be provided at geographically different location.
The processor should be intel quad core minimum 1.8 Ghz processor with
8 GB RAM,512 MB cache , RAID5 Dual hard disk 500GB minimum, with
dual Ethernet ports to enable redundant connection to LAN
OS shall be recent standard linux distribution.

I R I S E T
 ASTERIS
Asterisk is an open K
source software for implementation of IP-PBX
developed by DIGIUM corporation ,USA in 1999
It is designed for the linux OS and can be installed on either PC servers
or compatible embedded hardware.
It is free downloadable GPL (Gnu Public License) open source
software, can be down loaded from WWW.asterisk.org site.
The software includes source code and can be modified according to
user requirements under the terms of the GPLv2 license

I R I S E T
 FEATURES OF
ASTERISK
Various modern features available in asterisk
Call features: automated attendant, blind transfer, CDR , call forward,
Conference bridging, SMS messaging, streaming media access, talk
detection, voice mail , etc.
Computer –telephony Integration
Allows Scalability
Audio Codec Support (in standard Distribution)
Traditional telephony protocols support (with add-on hardware
ISDN protocols support (with add-on hardware)

I R I S E T
IP Exchanges which are purely based on software programming
and don’t use any hard modules instead they use all soft modules
or software files are called as IP Telephony servers. In this the
hardware part such as subscriber and trunk gateways are kept
out side the server.

Today we will see installation of server with Linux version

operating system and PBX software application (ASTERISK
software)

We will also see the configuration of accounts , dialplan and IP

soft phone configuration.

I R I S E T
 IP IP
STM STM
SERVER SERVER
ETHERNE
LAN T
L2- LAN L2- Wi-
SWITCH SWITCH Fi
SIP Trunk LAN LA
N
IP-PRI G/Way IP-FXS
IP-
PRI30 on
Phone Mobile
STM
TDM
Exchange ANALOG
PHONES
2W Copper

IDF MDF 2wire

Copper I R I S E T
 L2- TDM
IP LA PRI30 PRI30
SWIT LA Exchang
SERVER N GateWay Trunk
CH N e
Here we can see a server connected to Switch and one PRI30 Gateway is also connected
to the switch. The PRI30 Gateway have one ethernet port and one E1 port. This E1 port
carry 30 PRI trunk connected from Any Digital Exchange with PRI interface card.
How to install a Linux Operating system on a windows PC.
Download Oracle Virtual Manager from internet and install it on Your Windows PC.
Download Linux Debian version from internet and keep it in your download folder.
Open VM Box and create a virtual machine. Set the HDD memory space approximately
15GB and RAM not less than 1GB.
After creating the machine, Go to setting and do network setting by selecting a LAN adapter.
Connect the LAN adapter in bridge mode.
Then go to setting>storage, and select the installation media from download folder.
Now start the machine and follow the instruction for installation.
For installation of Debian version, Take the help of installation lab sheet from iriset modules.
http://10.195.2.7/moodle/
I R I S E T
 User Accounts are The Dail-plan is for
created in respective Conference Bridge, IVRS,
‘ASTERISK’ software file depending upon Voice-mail are also created
can be CLI or GUI. the Protocol to be used for users.
FreePBX is GUI by the user device.
User Accounts are Automatic Call
version of ASTERISK. searched in the Distribution and Uniform
PBX Application databse by their call Distribution can be
Serve Server with ‘ASTERISK’ has account name, Host IP configured to control huge
Linux-OS and different configuration address, Domain flow of incoming calls in a
r
ASTERISK PBX Application files. Name, Dail-plan
Password and Call Center. Some of the
PBX ‘ASTERISK’ The is
Context.
created for each user. features of ACD and UCD.
installed Least recent
Other dialplan for
features and services Wrap-up time for Agents
are also created for Music-on-hold for callers
users. Agent login and logout
They can be commonly Agent pause and Unpause
used by individuals or Add and remove the agent
may be by group of from the Queue.
user. Join multiple queue
Recording of calls for an
Agent or Agent by its own.