Voice Over Internet
Protocol
What is it?
■ VoIP allows us to make telephone calls using a computer
network, over a data network like the Internet.
■ VoIP converts the voice signal from our telephone into a
digital signal that travels over the internet then converts
it back at the other end so we can speak to anyone with
a regular phone number.
■ VoIP may also referred to as IP Telephony.
Picture
Components of VoIP
■ This diagram illustrates how a home network
connects devices to the internet using a cable
modem and phone adapter.
■ The components of VoIP includes the following:
Cable Modem:
■ This device connects to your cable TV line and
provides internet access to your home network.
Phone Adapter:
■ This device connects to your phone line and allows
you to make phone calls over the internet (VoIP).
Personal Computers:
■ These devices connect to the cable modem and/or
phone adapter to access the internet and make
phone calls.
Telephone:
■This device connects to the phone adapter to make phone
calls over the internet.
Internet:
■This represents the global network of interconnected
computers that allows you to access information,
communicate with others, and much more.
In summary, this diagram shows how multiple devices in a
home can connect to the internet, enabling them to perform
various tasks like web browsing, email, and online
communication.
Other major components
■ The central call processor is a piece of hardware running
a specialized database/mapping program called a soft
switch.
■ The central processor in the VoIP system is a key
component in the design, greatly affecting the resulting
voice quality, feature set, and bill of materials.
■ A soft switch is a software-based device that manages
calls and streaming in Voice over IP (VoIP) networks.
■ It establishes, maintains, routes, and terminates sessions in
VoIP networks. It also manages voice, fax, data, and video
traffic.
■ The soft switch know:
■ Where the endpoint is on the network.
■ What phone number is associated with that endpoint.
■ The current IP address assigned to that endpoint.
■ If a soft switch does not have the information, the request
■ A soft switch is a software-based device that manages
calls and streaming in Voice over IP (VoIP) networks.
■ It establishes, maintains, routes, and terminates
sessions in VoIP networks. It also manages voice, fax,
data, and video traffic.
■ The soft switch know:
■ Where the endpoint is on the network.
■ What phone number is associated with that
endpoint.
■ The current IP address assigned to that endpoint.
■ If a soft switch does not have the information, the
request is handled by another soft switch.
Protocols Used
There are various protocols involved to ensure that VoIP
technology functions properly, the main ones being:
Session Initiation Protocol (SIP):
It is the main multimedia communication protocol currently in
use. A way of connecting several callers, it is mostly used for
VoIP but is also found in online video games and instant
messaging services;
H.323: It is traditionally used to provide audio, visual and data
communication over the IP network, this protocol is valued
because it is easy to configure and use. However, H.323 is being
phased out and replaced by SIP, mainly because it is a relatively
static technology. That explains why not many
telecommunications devices are still compatible with it.
VoIP PSTN
Voice over a Internet Protocol uses the packet Public Switched Telephone Network uses circuit
switching. switching.
In VoIP, only an internet connection is In PSTN, dedicated lines required from telecom
required. company.
No dedicated path required between sender Dedicated path required between sender and
and receiver. receiver.
One VoIP lines require 100 kbps. One link typically supports 64 kbps.
In VoIP, it acquires and releases bandwidth as Bandwidth is reserved in advance it requires as each
it is needed. line is 64 kbps.
In VoIP, Cost is not dependent on distance and
In PSTN, Cost is based on distance and time.
time.
Extensions is a standard feature of VoIP. Extensions requires extra costs in PSTN.
VoIP provides consistent voice quality but
PSTN provides consistent voice quality.
depends on bandwidth.
The Call waiting, call forwarding and call transfer are
The Call waiting, call forwarding and call
usually the standard features in a PSTN but
transfer are the standard features of a VoIP.
sometimes it costs the extra for this.
Scalability and upgrades requires improved Scalability and upgrades requires substantial
internet bandwidth. hardware additions.
Other important protocols
Real-Time Transport Protocol (RTP)
Transmits audio and video packets in real time between
endpoints during a VoIP call. RTP uses time stamps and
ordinal numbers to identify packages.
Features
■ Multicasting
■ Payload type identification
■ Time shaping
■ Sequencing
■ Delivery monitoring
Real-Time Control Protocol (RTCP)
■Accompanies RTP and provides performance metrics while
helping with media stream synchronization.
■Provides feedback on the quality of data using lost packet
counts.
■Identifies and keeps track of participants.
■Sends retransmission requests.
A session generally consists of an RTP/RTCP pair of channels.
That usually works over UDP/IP.
RTP/RTCP
■ RTP Issues
■ No QoS guarantees
■ No guarantee of packet delivery
■ RTP Timestamp (TS) and Sequence Number (SN)
■ TS used to order packets in correct timing order.
■ SN to detect packet loss.
■ For a video frame that spans multiple packets – TS is
same but SN is different.
RTP/RTCP
Media Application
RTCP
RTP
UDP
IP
RTP/RTCP
RTP HEADER
The header format of RTP is very simple and it covers all real-
time applications.
■Version : This 2 bit field defines version number. The current
version is 2.
■P : The length of this field is 1 bit. If value is 1, then it denotes
presence of padding at end of packet and if value is 0, then
there is no padding.
■X : The length of this field is also 1 bit. If value of this field is
set to 1, then its indicates an extra extension header between
data and basic header and if value is 0 then, there is no extra
extension
■Contributor count : This 4 bit field indicates number of
contributors. Here maximum possible number of contributor is
15 ( as a 4 bit field can allows number from 0 to 15).
■M : The length of this field is 1 bit and it is used as end marker
■ Payload types :This field is of length 7 bit to indicate type of
payload.
■ Sequence Number: The length of this field is 16 bits. It is
used to give serial numbers to RTP packets. It helps in
sequencing. The sequence number for first packet is given a
random number and then every next packet’s sequence
number is incremented by 1. This field mainly helps in
checking lost packets and order mismatch.
■ Time Stamp: The length of this field is 32-bit. It is used to
find relationship between times of different RTP packets. The
timestamp for first packet is given randomly and then time
stamp for next packets given by sum of previous timestamp
and time taken to produce first byte of current packet.
■ Synchronization Source Identifier: This is a 32 bit
field used to identify and define the source. The value
for this source identifier is a random number that is
chosen by source itself. This mainly helps in solving
conflict arises when two sources started with the same
sequencing number.
■ Contributor Identifier: This is also a 32 bit field used
for source identification where there is more than one
source present in session. The mixer source use
Synchronization source identifier and other remaining
sources (maximum 15) use Contributor identifier.
Advantages of VoIP
■Cost
■Free VoIP to VoIP
■Low cost VoIP to Public Switch Telephone
Network (PSTN)
■Less bandwidth requirements
■Low cost / no cost software and hardware
■Mobility
■Any internet connection
■Growing number of wireless broadband
locations
Drawbacks
■Quality
■High quality PSTN
■Variable VoIP dependent on connection
■Dependent on wall power
■Lost or delayed packets cause drop-out
in voice
■Emergency Calls
■Hard to find geographic location
■Security
■Most VoIP services do not support
encryption