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What is VoIP

It is a form of communication
Can make phone calls over broadband internet connection
Can make calls over traditional landline numbers
Uses IP to transmit voice as packets over IP n/w
Voice digitized, compressed and converted to IP packet

VOIP configurations

Requirements, availability, limitations

Main issues
How to protect against risks
H.323 standard
Session Initiation Protocol(SIP)
Supporting protocols

VoIP configurations

Dedicated routers
Traditional phones are connected to cable/DSL modems.
Once configured, it require no special software
Can use traditional phones
It is slightly larger than normal USB
Software controlled VoIP
Softphone software
Need a computer with necessary equipements
Less cost
Dedicated VoIP phone
It connects directly to a computer network
Consist of a phone and a base station that connects to the internet

Requirements, Availability and


A connection to the internet through ISP
VoIP service to reach traditional landlines
VoIP software
Need power source
Need high speed broadband internet connection
SPIT(SPam over Internet Telephony)- telemarketing
Spoofing- an attacker can masquerade as another VoIP caller
Confidentiality- VoIP data travels unencrypted over internet

How to protect against risks

Use and maintain anti-virus and anti-spyware programs.
Be cautious about opening files attached to email messages or instant
Verify the authenticity and security of downloaded files and new software.
Configure your web browser(s) securely.
Use a firewall.
Identify, back-up, and secure your personal or financial data.
Create and use strong passwords.
Patch and update your application software.
Do not divulge personal information to people you dont know.
If you are using a software VoIP application, consider using encryption
software for both your installation and for those you wish to talk to.

Main issues

Quality of Service
IP was designed for carrying data
VoIP can provide best effort service only
It needs to work with different vendor in a public network environment
Anyone can capture the packets in the data since it is in plain format
Integration with PSTN
It need to cooperate with Public Switched Telephone Network
VoIP may grow to large user market

H.323 standard
Components of H.323
H.323 protocol stack
Call setup in H.323
Components of SIP
SIP messages
SIP operation
H.323 vs SIP

ITU standard for VoIP communication (International
Telecommunication Union)
No QoS is provided
Multipoint Control unit

Endpoints which makes a call
Should support some protocols
H.245 for allowing the usage of channels,
Q 931- for call signaling and setting up call
RTP(Real time Transport Protocol)- carries voice packets
RAS(Registration, Admission Status)- to interact with gate
Interface between LAN/endsystem with PSTN
They perform translation b/w different formats
Put voice data from PSTN in public n/w and vice versa

acts as a manager
Central point for all calls from its zone

Address Translation
Admissions Control
Call signaling
Call Authorization
Bandwidth Management
Call Management

Multipoint Control Units (MCU)

provides the capability for three or more terminals and
gateways to participate in a multipoint conference
Multipoint Controller and Multipoint Processor

H.323 Protocol Stack

Call Setup in H.323

Discovering a gatekeeper
Discovers the gatekeeper to which the endsystem shoutld register
Multicasts Gatekeeper Request message(GRQ)
Replies with either
Gatekeeper Confirmation (GCF) message with its transport address
Gatekeeper Reject(GRJ) message

Endpoint may choose any one gatekeeper

If timeout, retrasmits the request
Registration of the endpoint with its gatekeeper
Endpoint sends Registration request(RRQ) to gatekeeper
Gatekeeper sends Registration Confirmation(RCF) or Registration
Unregister Request(URQ) endpoint/gatekeeper can cancel its request
Unregister Confirmation(UCF)

Endpoint enters the call setup phase

To determine contact information
Issue Location Request(LRQ) message
Location Confirmation(LCF) with contact information

The capability exchange

To provide admission control and bandwidth management functions
Admission Request(ARQ)
Admission Confirm(ACF)
Bandwidth Change Request(BRQ)

The call is established

After call establishment, media control protocol will be executed

Determining master/slave
Capability exchange
Media Channel Control
Conference Control

When the endpoint is done, it can terminate the call.

SIP(Session initiation
IETF standard for VoIP
Application layer control protocol
Client sends request, server process
the request
SIP depends on Session Description

User agent : end system for client , server
Network servers
Registration Server- updates current locations of user
Proxy server- forwards request to next hop server
Redirect server- determines the next-hop server and returns the
address of next-hop to client instead of forwarding
INVITE: for inviting a user to a call
BYE: for terminating a connection between the two end points
ACK: for reliable exchange of invitation messages
OPTIONS: for getting information about the capabilities of a call
REGISTER: gives information about the location of a user to the SIP
registration server.
CANCEL: for terminating the search for a user

SIP Operation

SIP Addressing
SIP hosts are identified by a SIP URL
Locating a SIP server
client can send the request to a SIP proxy server
SIP Transaction
A request together with the responses triggered by that request make up
a SIP transaction.
SIP Invitation
The INVITE request asks the callee to join a particular conference
Callee response by sending an ACK request
Locating a User
locations can be dynamically registered with the SIP server
Changing an Existing Session
This is done by re-issuing the INVITE message

H.323 vs SIP