(Voice Over Internet Protocol)

By samresh chand

Introduction
VoIP allows phone calls to be made by using Broadband Internet connection. Also known as Internet Phone. Distance doesnt matter. Contain all normal phone features. First Gen. Phone : PingTel Expressa (since 2001) Second Gen. Phone :Cisco System (since 2002)

VoIP digitalize voice data in data packets sends Them & reconvert them in voice at destination.

Voice

Source

ADC

Internet

DAC

Voice

Dest.

Some service provider : SunRocket Skype Webphone

System Requirement
Processor 1GHz or higher RAM 64 MB or higher Microphone & speaker or headset or handset OS window 98/2000/XP

Network Requirement
DSL (Digital subscriber line)
Cable Modem

Flavors of VoIP
ATA :(Analog telephone adaptor) Simple & most common way It allows to connect std. phone to computer ATA converts analog signal of traditional phone into digital data

IP phone : Specialized phone but looks like normal Have RJ-45 Ethernet connector Connected directly to router

Computer-to-Computer : Easiest way to use Needs s/w,microphone,speaker

Protocols :
SIP (Session initiation protocols) MeGaCo (Media gateway control protocol) H.323

SIP (session initiation protocol)

Developed by IETF MMUSIC working group Standard for initiating ,modifying,& terminating interface user session that involves multimedia Peer-to-peer In nov 2000 SIP was accepted as 3GPP signaling protocol & permanent element of IMS architecture It describe media content of session e.g. IP port to use code being used

Goal for SIP was to provide a superset of all call processing function & features present in PSTN Session are simply packet stream of RTP where RTP is carrier for the actual voice or video content itself. SIP is similar to HTTP & shares some of its design principles & status code e.g. Human readable Request-response structured SIP is also used for Instant Messaging & Presence e.g. MSN messenger

Skype

SIP makes use of proxy servers to help route requests to the Users current location Authenticate & authorize users for services Implement provider call-routing policies Provide features to users

SIP also provides a registration function that allows users to upload their current locations for use by proxy servers.

MeGaCo
Media gateway control protocol is a signalling protocol used between a Media Gateway & a Media Gateway Controller in a VoIP network. MeGaCo is result of joint work of IETF & ITU. It is also known as H.248 by ITU. Later IETF gave MeGaCO

H.323
It is an umbrella recommendation from the ITU-T that defines the protocols to provide audio-visual communication on any network. It is called as Protocol of Choice due to its simplicity. It is like Fiber backbone & is best where everything is under the control of the network operator. Its popularity has decreased in the local loop due to its poor traversal of NAT & Firewalls.

Advantages
It is cheaper. No boundaries. Digital so compact & more noise tolerant. Uses internets packet-switching capabilities. Extra features (voicemail) Many flavors.

Hurdles
Less reliable.

Dependant on wall power.

Hacking.

Viruses
Emergency calls.

More initial cost.

Conclusion
The driving factor for most of todays VoIP offerings is lower cost for a conventional telephony service. However, using new, advanced technology for simply reducing prices, in a market that already is covered by existing services, will not allow stable and healthy market players. Instead, the total amount of money earned will decrease and will have to be shared by a growing number of market players. Costs do need to be reduced, but customers should also expect innovative new products and features.

Any