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VoIP

PRESENTATION
By U D Dalal
Distributed Multimedia
Systems (DMS or DMMS)
A DMS is an integrated communication, computing,
and information system that enables the processing
management, delivery and presentation of
synchronized multimedia information that the
quality of services gurantees.
Broadly classified in the three
distributed multimedia
applications
 Interactive Television (ITV)—Video programs
and interact with them
 Telecooperation—remote learning,
telecommuting, tele servicing, teleoperation,
multimedia e-mails, videophone, desktop
conferencing, electronic meeting rooms, joint
editing and groupdrawing
 Hypermedia (link to other multimedia)—
home-shopping, interactive videogames,
financial transaction, video on demand (VOD),
news on demand, digital libraries, electronic
encyclopedias, multimedia magazines,
information kiosks, computer aided learning
tools, the WEB
 Wireless Printers
 All Interactive applications
Distributed Multimedia System Example
Challenges in DMS

 Real time delivery of audio and video data and


constraints of bit error rate for the text data
 Packet loss probability and delivery delays, latency and
zitter, out-of-order delivery or misdelivery of packets
 Synchronization among multiple media streams from
distributed sources for meaningful presentation
 Round trip time must be less than 250 ms.
 Priority scheduling scheme requirement
MOTIVATION
What is VoIP
• It is INTERNET PROTOCOL TELPHONY (VOICE over IP)
Transmission of voice telephone calls using internet infrastructure.

• IP Telephony must be backward compatible with existing PSTN.


PSTN ( Public Switched Telephone
Network)

 Known structure of conventional telephone network.

İstanbul, yeditepe Ankara, METU


PSTN

Call Direction

Since the telephone was invented


in the late 1800s, telephone communication has not changed
substantially
A Change of Perspective

Reality
Web

Voice
Video
Data
Email File
Transfer
What is it?
 VoIP allows you to make telephone calls using a
computer network, over a data network like the
Internet.

 VoIP converts the voice signal from your telephone


into a digital signal that travels over the internet
then converts it back in analog form at the other end
so you can speak to anyone with a regular phone
number.

 When placing a VoIP call using a phone with an


adapter, you'll hear a dial tone and dial just as you
always have. VoIP may also allow you to make a call
directly from a computer using a conventional
telephone or a microphone.

 Voice conversations are turned into digitized data and


packetized for transmission across a network.
Why need to IP Telephony?
 Economic (uses internet, IP routers...)
 Further savings. Because underlying network
infrastructure can be shared. A single set of equipment,
wiring, network connection enough.
 Not only voice but also video is transmitted using
similar concepts.
 Spreads at a fast pace. For instance there isn’t any
internet cafe which doesn’t have tiny video cameras
over their monitors.
 Independent. Much larger selection of service
providers to provide voice and video communication
services . No geographical restriction! Located virtually
anywhere in the world!
VoIP to VoIP

Broadband Network

VoIP VoIP

Call Direction

IP Protocol
VoIP to POTS with Internet

İstanbul, yeditepe
Internet/Broadband
New Delhi
PSTN

VoIP
Server/Gateway
India
Call Direction

IP Protocol
VoIP to POTS without Internet

Server/Gateway
İstanbul, kayışdağı Miami, FL
PSTN

VoIP CO LATA Miami

Call Direction

IP Protocol
VoIP Conversion steps at
physical layer
 Continuously sample audio.
 Convert each sample to digital form.
 Send the resulting digitized stream accross an IP
network in packets.
 Convert the stream back to analog for playback.
 Before the procedure above, the system must handle
call setup. Ph number to IP conversion and later reverse
conversion.

Two groups have introduced signaling standards for IP telephony


International Telecommunications Union(ITU), controls telephone standards.
Internet Engineering Task Force(IETF), controls TCP/IP standards.
ITUH.323
IETFSession Initiation Protocol (SIP)

Aboves must be able to interact with SS7


IP address to Phone Number

 VoIP look for IP address


 Translate Phone numbers to IP addresses
 The central call processor is a piece of
hardware running a specialized
database/mapping program called a soft
switch.
 Soft switches 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 soft switch does not have the information,
the request is handled by another soft switch.
Protocols
 Used to connect different pieces of hardware.
 H.323
 Most widely used protocol
 provides specifications for real-time, interactive
videoconferencing, data sharing and audio applications
(VoIP)
 SIP
 More streamlined protocol
 Developed specifically for VoIP
 Lack of a standard protocol is a problem. Not
always compatible.

 Use of UDP and RTP


 Encoding Pulse Code Modulation (PCM)
 TransmissionReal-Time Transport Protocol
(RTP)
Note: RTP is not a tranport-layer protocol. It is a tranfer
protocol that operates at Layer 5.

 Each RTP message is encapsulated in UDP


datagram, which is then encapsulated in an IP
datagram for transmission.
Why UDP instead of TCP?

 Higheroverhead of TCP
does not make sense for
telephone call. Because
audio must stream! No wait
for missing packets. Play
missing part as silence.
Why UDP instead of TCP?

 UDP Offerrs best-effort delivery. to handle duplication,


delay, out-of-order delivery, each RTP message
contains:
 A sequence number
 A real-time clock value
What does real-time clock
value do?
 Allows a receiver to construct the axact temporal
sequence of the data.
 İf a packet is missing , the receiver knows exactly how
long to wait before starting to play the next packet
Signaling Systems and
Protocols
 Signaling: The process of establishing and terminating a
call. Includes:
 Mapping a phone number to location
 Finding a route to the called party
 Handling other details such as call forwarding
 Signaling System 7 (SS7) for traditional telephone
system.
A Basic IP Telephone System

Two basic components interconnected by an IP internet.


*IP telephone
*Media Gateway Controller

Media Gateway Controller

IP telephone IP telephone
INTERNET
Media Gateway Controller Media gateway handles
voice

IP telephone

Analog
INTERNET
PSTN telephone

Signaling gateway
handles setup

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