1. VoIP
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2. VoIP System

3. Gatekeeper: A gatekeeper is useful for handling VoIP call
connections includes managing terminals, gateways and MCU's
(multipoint control units).
A VoIP gatekeeper also provides address translation, bandwidth
control, access control
A VoIP gatekeeper can improve Security and Quality of Service
(QoS)
VoIP Gateway: A VoIP gateway is also required to handle
external calls. A VoIP gateway functions as a converter that
converting VoIP calls to/from the traditional PSTN lines
VoIP Clients: Other required VoIP hardware includes a VoIP
client terminal, a VoIP device could be an IP Phone, or a
multimedia PC or a VoIP-enabled workstation runs VoIP
software.

4. PSTN vs. VoIP PSTN VoIP VoIP uses packet switching.
No dedicated path between sender and receiver.
It acquires and releases bandwidth, as it is needed.
Cost is not dependent on time and distance.
Features such as call waiting,
Caller ID and so on are usually
included free with service
Voice networks use circuit
switching.
Dedicated path between
calling and Called party.
Bandwidth is reserved in
advance. Each line is
64kbps.
Cost is based on distance
and time.
Features such as call
waiting, Caller ID and so on
are usually available at an
extra cost

5. PSTN VoIP
Can be upgraded or expanded
with new equipment .
Long distance is usually per
minute or bundled minute
subscription.
Hardwired landline phones
(those without an adapter)
usually remain active during
power outage.
Upgrades usually requires only bandwidth and software upgrades.
Long distance is often
included in regular monthly price.
Lose power, lose phone
service without power backup in place.

6. Basic Principles of VoIP
Audio Codec, Video Codec
Data Transport (RTP, RTCP)
Addressing
Signaling (SIP, H.323)

7. Audio Codecs Are used to convert analog signal into digital data.
The most common codecs for VoIP are
Codec Bandwidth/kbps G
G /56/64
G /6.3
Stands for coder-decoder:
Since voice contains lot of data, it is compressed by coders without
compromising the reliability and quality of voice signal.

8. Translation of Analog Signal to Digital Signal

9. Video Codec Video Codec: common examples include H.261
(for 64kbps and above), H.263 (for 64kbps and
below), and MPEG 4.
The encoded information is then encapsulated
within an IP packet and these packets are then
transported across the network to their destination.

10. Data Transport (RTP,RTCP)
It stands for Real time Transport Protocol.
Application layer protocol for transmitting real time data
(audio, video, ...)
Includes sequence numbering, time stamping, delivery
monitoring.

11. RTCP It stands for Real time Transport Control Protocol.
While RTP carries the media streams (e.g., audio and
video), RTCP is used to monitor transmission statistics
and quality of service (QoS) and aids synchronization
of multiple streams.
Main functions:
support for multi-point communication

12. VoIP SIGNALING PROTOCOLS
Signaling in VoIP is needed for
-To establish a point to point connection and to keep it
open for the duration of the call.
-Agreeing on coding /decoding procedures.
Types of Signaling Protocols:
H.323
SIP

13. H.323

14. H.323
H.323 is a set of protocols for sending voice, video and data over IP network
to provide real-time multimedia communications. H.323 is reliable and easy
to maintain technology
1. Terminals
2. Gateways
3. Gatekeepers
4. Multipoint control units (MCUs)
An H .323 zone is a collection of all terminals, gateways, and MCUs managed
by a single gatekeeper. A zone includes at least one terminal and may include
gateways or MCUs. A zone has only one gatekeeper. A zone may be independent
from network topology.

15. H.323 …… There are four basic entities in a default H.323 network :
Terminal: H.323 terminal also called H.323 client is the end-user device. It could be
IP telephone or a multimedia PC with another H.323 client. That provides real-time
two-way media communication.
Gateways (GW): A Gateway (GW) is an optional component that provides inter-net
work translation between terminals.
Gatekeepers (GK): A Gatekeeper (GK) is an optional component provides address
translations and access control services.
Multipoint Control Units (MCU):A Multipoint Control Unit (MCU) functions as a
bridge or switch that enables three or more terminals and gateways in a multipoint
conference.

16. H.323 Characteristics
Allows transmission of video and data while a phone call is in progress
Incorporates protocols for security.
Uses a special hardware Multipoint Control Unit for conferencing calls.
Defines servers for address resolution,
authentication, accounting, features, etc.

17. SIP SIP stands for Session Initiation Protocol.
Developed by IETF since 1999.
SIP is the core protocol for initiating, managing and
terminating communication sessions (i.e audio &
video call) over the Internet.
These sessions may be text, voice, video or a
combination of these.
SIP sessions involve one or more participants and can use
unicast or multicast communication.
Sessions include Internet Multimedia conferences or Internet
Telephone calls.

18. SIP SIP is a signaling control protocol which is similar to http.
it’s designed to initial and terminate
VoIP sessions with one or more
participants.
It is less weight and more flexible
than H.323 that also can be used for
multimedia sessions such as audio,
video and data.

19. SIP SIP has two components: User Agents and SIP servers.
User agents are peers in a SIP. User agents could be either an agent client
or an agent server. A user agent client initiates by sending a SIP request.
A user agent server can accept, terminate or redirect the request as
responses to this SIP request.
There are three types of SIP servers include SIP proxy servers, SIP
registrar servers, and SIP redirect servers.
A SIP server functions as a server that handles these requests, e.g. requests
transferring, security, authentication, and call routing.
e.g. Microsoft MSN Messenger, Apple iChat.

20. SIP Characteristics Operates at the application layer.
Encompasses all aspects of signaling, e.g. location of called
party, ringing a phone, accepting a call, and terminating a call.
Provides services such as call forwarding.
Relies on multicast for conference calls.
Allows two sides to negotiate capabilities and choose the media
and parameters to be used.

21. SIP Methods
Six basic message types, known as methods:

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