Fundamental of VoIP

Muhammad Fayzan Siddiqui

Bahria university Karachi

Asad Ahmed Singani

Bahria university Karachi

Abstract
Voice Over internet protocol is a technique in which
ordinary voice signals are converted to packets and
send over IP networks. VoIP is very cheap way to
transfer voice over a long distance. In 1995 Vocal Tec
Company of Israel started Internet telephone software
as “Internet Phone Since then in just four year it has
grown very rapidly. In modern era the VOIP is being
used to rout the international calls.
Keywords: VOIP, Voice Over Internet Protocol,
Packets, Switching, Gateways.
1. Introduction:
VoIP is sometimes referred to as Internet telephony; it
is a method of digitizing voice, into packets. Headers
are added to the packets and transmitting those packets
over a packet switched IP network. [2]
3. Types of Switching
There are two type of switching
3.1. Circuit Switching:
In circuit switching there is a dedicated communication
path between two stations. There are three phases call
establishment, maintenance and termination. In circuit
switching there must have switching capacity and
channel capacity to establish connection.
3.2. Packet Switching:
In packer switching, single node to node link can be
shared by many packets over time. Packets are accepted
even when network is busy. Packets are handled in two
ways virtual circuit and datagram.

“Fig1.1 Rough Conceptualization of VoIP”

2. Internet Protocol:

IP itself is a connectionless protocol that resides at

Layer 3 (the network layer)
It is unreliable. Other protocols, such as TCP, can sit on
“Fig 1.2 Circuits vs. Packets”
top of IP (Layer 4, session) and can add flow control, Source: Peter Ingram, “Voice over Internet Protocol- An
sequencing, and other features. introduction”, OfCom
Protocols split transmitted data into packets, add
necessary addressing information to the packets and
transmit them and assemble again data in receiving end. 4. Overview of PSTN:
[3]
The telephone infrastructure starts with a simple pair of
copper wires running to the subscriber, known as a local
loop. The local loop physically connects subscriber
telephone to the central office switch or exchange.
The communication path between several central office
switches is known as a trunk. (e.g. E1, T1)
Switches are currently deployed in hierarchies. End
office switches (or central office switches) interconnect
through trunks to tandem switches (also referred to as
Class 4 switches). Higher-layer tandem switches
connect local tandem switches.
Central office switches often directly connect to each
other. Where the direct connections occur between
central office switches depends to a great extent on call
patterns. If enough traffic occurs between two central
office switches, a dedicated circuit is placed between
the two switches to offload those calls from the local
tandem switches. Some portions of the PSTN use as
many as five levels of switching hierarchy. [1]
“Fig1.3. PSTN—Traditional VoIP”
Source: Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom
5. PSTN Signaling:
Generally, two types of signaling methods run over
various transmission media. The signaling methods are
broken into the two groups:
• User-to-network signaling— this is how an end user
communicates with the PSTN.
• Network-to-network signaling— this is generally how
the switches in the PSTN intercommunicate.
Network-to-network signaling also uses an out-of-band
signaling method known as Signaling System 7 (SS7)
(or C7 in European countries). [1]
6. PSTN Services and Applications:
The popular custom calling features commonly found in
the PSTN today:
• Call waiting—Notifies customers who already
placed a call that they are receiving an incoming
call.
• Call forwarding—Enables a subscriber to forward
incoming calls to a different destination.
• Three-way calling—Enables conference calling.
With the deployment of the SS7 network, advanced
features can now be carried end to end. A few of
the CLASS features are mentioned in the following
list:
• Display—Displays the calling party's directory
number, or Automatic Number Identification
(ANI).
• Call blocking—Blocks specific incoming numbers
so that callers are greeted with a message saying
the call is not accepted.
• Calling line ID blocking—Blocks the outgoing
directory number from being shown on someone
else's display. (This does not work when calling
800-numbers or certain other numbers.)
• Automatic callback—enables you to put a hold on
the last number dialed if a busy signal is received,
and then place the call after the line is free.
• Call return (*69)—Enables users to quickly reply
to missed calls. [1]
7. Drawbacks to the PSTN:
Although the PSTN is effective and does a good job at
what it was built to do (that is, switch voice calls), many
business drivers are striving to change it to a new
network, whereby voice is an application on top of a
data network. This is happening for several reasons:
• Data has overtaken voice as the primary traffic
on many networks built for voice.
• The PSTN cannot create and deploy features
quickly enough.
• Data/Voice/Video (D/V/V) cannot converge on
the PSTN as currently built.
• The architecture built for voice is not flexible
enough to carry data. [1]
8. VoIP Technical Description:
Figure 1.5 shows a simplified block diagram of VoIP
operation from an analog signal deriving from a
standard telephone, which is digitized and transmitted
over the Internet via a conversion device. Then, at the
distant end, it is converted back to analog telephony “Fig1.5. PC-to-PC VoIP”
using a similar device suitable for input to a standard Source: Peter Ingram, “Voice over Internet Protocol- An
telephone. The gateway is placed between the voice introduction”, OfCom
codec and the digital data transport circuit. An identical
device will also be found at the far end of the link. This 9.2. Phone-to-Phone:
equipment carries out the signaling role on a telephone
call among other functions. If the traditional telephone is use it will connect to an
Moving from left to right in Figure 1.5, we have the Analogue Telephone Adapter (ATA) or use an IP
spurty analog signal deriving from a standard telephone phone. The called party may be another VoIP user or,
set. The signal is then converted to a digital counterpart via a gateway, a traditional PSTN customer. [6]
using one of seven or so codecs [coder-decoder(s)] that
the VoIP system designer has to select from.
The binary output of the codec is then applied to a
conversion device (i.e., a “packetizer”) that loads these
binary 1s and 0s into an IP payload of from 20 to 40
octets in length.
The output of the converter consists of IP packets1 that
are transmitted on the web or other data circuit for
delivery to the distant end.
At the far end the IP packets or frames are input to a
converter (i.e., depacketizer) that strips off the IP
header, stores the payload, and then releases it in a “Fig1.6. Phone-to-Phone VoIP”
constant bit stream to a codec (i.e., a D–A converter). Source: Peter Ingram, “Voice over Internet Protocol- An
Of course this codec must be compatible with its near- introduction”, OfCom
end counterpart. The codec converts the digital bit
stream back to an analog signal that is then input to a 10. VoIP in PSTN:
standard telephone subset. [2]
Many traditional PSTN calls are carried as VoIP in part,
for efficiency reasons they may travel with other IP
traffic. These calls are different from other VoIP calls.
It is invisible to end customer. The private IP network is
used instead of internet cloud to control quality. [6]
“Fig1.4. Conceptualization of VoIP”

9. VoIP Calls:
Types of VoIP calls

9.1. PC-to-PC:

Software install on the machine (PC) allows voice “Fig1.7. VoIP in the PSTN”
“calls” from one PC to another. It enables voice to Source: Peter Ingram, “Voice over Internet Protocol- An
convert into IP packets at PC. [6] introduction”, OfCom

11. Function of Gateway:

 Gateway
Internet
PSTN
“Fig1.8. Concept of Gateway”
Source: Roger L. Freemen, “Fundamentals
Telecommunication, IEEE press,
Gateway is a server; it may also be called a media
gateway. It is the bridge of Internet and PSTN to
connect. For PSTN the Gateway has the interface
of telephone net. (e.g. E1 (2048 kb/s)), and for
Internet side, Gateway has network interface (E.g.
Ethernet interface). Through the cooperation of
hardware and software, realize between" Circuit
switch" and "Packet switch" voice coding form
change mutually.
Media gateways are part of the physical transport
layer. They are controlled by a call control function
housed in a media gateway controller. It supports
several types of access networks including media
such as copper, fiber, radio, and CATV cable. [2]
12. VoIP Protocols:
Protocols are set of rules determining the format
and transmission of data, so protocols will
determine that what will be the format of the data
and how will be data transmitted. [4]
The most common VoIP protocols used are:
1. Megaco H.248
2. MGCP
3. Media Gateway Control Protocol
4. MIME
5. RVP over IP
6. Remote Voice Protocol Over IP Specification
7. SAPv2
8. Session Announcement Protocol
9. SDP
10. Session Description Protocol
11. SGCP
12. Simple Gateway Control Protocol
13. SIP
14. Session Initiation Protocol
15. Skinny
16. Skinny Client Control Protocol (SCCP)
17. Gateway Control Protocol
12.1. Media Gateway Control Protocol:
MGCP is an implementation of the Media Gateway
Control Protocols architecture for controlling Media
Gateways on Internet protocols networks and the public
switched telephone network. MGCP is a signaling and
call control protocol used within VoIP systems that
typically interoperate with the public switched
telephone network. Media Gateway Control Protocol is
used between elements of a decomposed multimedia
of gateway which consists of a Call Agent, which contains
the call control "intelligence", and a media gateway
which contains the media functions, e.g., conversion
from TDM voice to Voice over IP. [4]
12.2. Multipurpose Internet Mail Extensions:
MIME is a specification for enhancing the capabilities
of standard Internet electronic mail. It offers a simple
standardized way to represent and encode a wide variety
of media types for transmission via Internet mail.
The MIME standard contains the following types of
messages:
• Text messages in US-ASCII
• Character sets other than US-ASCII.
• Multi-media: Image, Audio, and Video
messages.
• Multiple objects in a single message.
• Multi-font messages.
• Messages of unlimited length.
• Binary files. [4]
12.3. Remote VoIP:
Remote Voice Protocol (RVP) is MCK (McKesson
Corporation) Communications' protocol for transporting
digital telephony sessions over packet or circuit based
data networks. The protocol is used primarily in Mack’s
Extender product family, which extends PBX services
over Wide Area Networks (WANs).
RVP/IP uses TCP to transport signaling and control
data, and UDP to transport voice data. Control and
signaling packets carried over TCP are encapsulated
using the following format; a header followed by
signaling or control messages: [4]
“Fig1.9. RVP over IP packet structure”

12.4. Session Initiation Protocol (SIP):
Session Initiation Protocol (SIP) is an application layer
control simple signaling protocol for VoIP
implementations using the Redirect Mode. SIP is a
textual client-server base protocol and provides the
necessary protocol mechanisms so that the end user
systems and proxy servers can provide different
services.
Call forwarding in several scenarios: no answer, busy,
unconditional, address manipulations (as 700, 800, and
900- type calls).
Called and calling number identification
Personal mobility, caller and called authentication,
invitations to multicast conference, basic Automatic
Call Distribution (ACD).
SIP supports five distinct features of establishing and
terminating multimedia communications: User location,
User capabilities, User availability, and Call setup, Call
handling. [4]
12.5. T.38
The T.38 IP-based fax service maps the T.30 fax
protocol onto an IP network. Both fax and voiced data
are managed through a single gateway. T.38 uses 2
protocols, one for UDP packets and one for TCP
packets. Data is encoded using ASN.1 (Abstract Syntax
Notation One) to ensure a standard technique
It allows users to transfer facsimile documents between
2 standard fax terminals over the Internet or other
network using IP protocols. [4]
12.6 The Real-time Transport:
The Real-time Transport (RTP) Protocol provides end-
to-end network transport functions suitable for
applications transmitting real-time data such as audio,
video or simulation data, over multicast (the delivery of
information to a group of destinations simultaneously)
or unicast (the sending of information packets to a
single network destination.) network services.
RTP does not address resource reservation and does not
guarantee quality-of-service for real-time services. The
data transport is augmented by a control protocol “Real
Time Transport Control Protocol” to allow monitoring
of the data delivery. [4]
13. Voice over IP Benefits and Applications:
• Benefits of Voice over IP (VoIP), including
cost savings, single infrastructure savings, and
new applications
• Using a packet telephony call center versus a
circuit-switched call center
• Service provider prepaid calling card
applications
• Service provider enhanced services (such as
Internet call waiting and click to talk) [2]
14. VoIP in Pakistan:
Voice over IP is nothing new in Pakistan from - VOIP
has played an important role for people to stay close to
those who are far away from them. In the current times
when life is becoming more mobile, keeping oneself
attached to PC, is very cumbersome and the increasing
power crises in the country makes it more difficult to
use it.
Here comes VOIP on mobile, supported by SKYPE and
Fring. FRING is free software which supports to use
SKYPE with ease, just register once and you can talk
away any where you are - all you need is a supporting
cell phone and data cable connection.
Fring is versatile, it’s not limited only to high end
phones, from Nokia 1680 to Nokia N97, from Symbian
to UIQ to Windows Mobile and iPhone, and it works on
every kind of cell phones effortlessly. There are a lot of
advantages of VOIP, as international phone calls from
Pakistan is almost free but here is a need to pay charges
for calling to Pakistan. With Mobile VOIP, a
businessman can have his worldwide associates call him
over on his SKYPE id whether he is in office or
lounging at home, a family can talk away via speaker
phone with their loved ones abroad even if the lights are
out, those concerned with security can sleep easy, it’s
rumored that only NASA has the technology to listen
over SKYPE and that too will take much of their
precious super computer time to hack into the
conversation. [5]
15. REFRENCES
[1] Jonathan Davidson, James peter “Voice over IP
Fundamentals”, Cisco press release, USA, 2000
[2] Roger L. Freemen, “Fundamentals of Telecommunication,
IEEE press, Canada, 2005
[3] www.wikepedia.com
[4] www.protocols.com
[5] www.propakistan.com
[6] Peter Ingram, “Voice over Internet Protocol- An
introduction”, OfCom, 18th January 2005