VOICE OVER INTERNET PROTOCOL (VoIP)

Presentation by, V.Nithya M.Tech.IT(Final Year) 1631010033

Overview
VoIP is voice over an Internet Protocol (IP) based network Voice over Internet Protocol, VoIP describes the category of hardware and software that enables people to make telephone calls via the Internet. Voice signals are converted to packets of data, which are transmitted on shared, public lines, hence avoiding the tolls of the traditional, public-switched telephone network (PSTN). VoIP uses the Internet Protocol (IP) to transmit voice as packets over an IP network.

Overview
VOIP can be achieved on any data network that uses IP, like the Internet, Intranets and Local Area Networks (LAN). The voice signal is digitized, compressed and converted to IP packets and then transmitted over the IP network. VoIP services need only a regular phone connection, while others allow you to make telephone calls using an Internet connection instead.

VoIP Architecture

How they fit in:The ISO Model

ISO Model Layer Presentation Session Transport Network Data Link Protocol or Standard Applications / CODECS H.323 & SIP RTP / UDP / TCP IP Non QOS ATM, FR, PPP, Ethernet

VoIP Protocols
H.323 Multimedia Standard
H.225 RAS - Registration, Admission, Status Q.931 - Call Signaling (Setup & Termination) H.245 - Call Control (Preferences, Flow Control, etc.) Lots of G.7XX CODECS for audio

SIP Session Initialization Protocol

H.323
Definition: a multimedia standard that provides a foundation to transport voice, video and data communications in an IP based non-QOS network. H.323 Zone
Collection of terminals, gateways, MCUs registered with a single gatekeeper.

H.323 Entities
Terminals (LAN Endpoints) Gateways (Optional but really useful) Gatekeepers (Also optional) MCUs

H.323 Equipment
Gateway
Device that connects H.323 voice network to non-H.323 voice network (SIP or PSTN) Allows H.323 terminals to communication with nonH.323 terminals

Gatekeeper
Provides address translation (H.323 & E.164 to IP) Admission control for H.323 terminals and gateways Manage bandwidth allocation Other optional services

H.323 Equipment
MCU (multipoint control unit)
MC multipoint controller Routes call and control signaling to ensure endpoint compatibility MP multipoint processor Switches, mixes and processes vice and video streams to conferencing equipment

Terminal
An endpoint that supports 2-way streaming with another H.323 terminal or gateway Originates and terminates calls Includes videoconferencing stations, hard phones, & soft phones

H.323 Call Setup via Gatekeepers

SIP
SIP Features for VoIP -Simplicity
-Scalability -Modularity -Internet-enabled

SIP in market -Many products but no deployment case

-Lack of multimedia applications and services

-SIP-SIP,SIP-H.323,SIP-PSTN and Intelligent network

services

SIP Request Messages

SIP Response Messages

100 Trying 180 Ringing 181 Call is being Forwarded 182 Queued

200 OK 301 Moved Permanently 302 Moved Temporarily

Example of SIP message

INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 167.180.112.24 From: sip:[email protected] To: sip:[email protected] Call-ID: [email protected] Content-Type: application/sdp Content-Length: 885 c=IN IP4 167.180.112.24 m=audio 38060 RTP/AVP 0 Notes: HTTP message syntax sdp = session description protocol Call-ID is unique for every call. Here we dont know Bobs IP address. Intermediate SIP servers will be necessary. Alice sends and receives SIP messages using the SIP default port number 5060.

Alice specifies in Via: header that SIP client sends and receives SIP messages over UDP

Comparison of Packet vs. Circuit Switching

Circuit
Call Setup Communications Channel Addressing Database / SS 7 Overlay Dedicated NANP

Packet
H.323 & SIP Shared IPv4 & IPv6

PC Configuration for VoIP

What it does.
VoIP can allow you to make a call directly from a computer, a special VoIP phone, or a traditional phone using an adapter to another VoIP phone or regular phone.

Transmission of Voice Using IP Networks

Step 1: Because all transmissions must be digital, the callers voice is digitized.
Step 2: Next using complex algorithms the digital voice is compressed and then Separated into packets; and using the Internet protocol, the packets are addressed and sends across the network to be reassembled in the proper order at the destination.

Transmission of Voice Using IP Networks

Step 3: During transmission on the Internet, packets may be lost or delayed, or errors may damage the packets. Conventional error correction techniques would request retransmission of unusable or lost packets, but if the transmission is a real-time voice communication that technique obviously would not work, so sophisticated error detection and correction systems are used to create sound to fill in the gaps. Step 4: After the packets are transmitted and arrive at the destination, the transmission is assembled and decompressed to restore the data to an approximation of the original form.

Internet Telephony
The use of the Internet that was originally designed to carry computer data to carry voice -A packet switched network Voice was originally carried over circuit switched networks -PSTN

Connecting to regular phone

Switch from digital to analog

To make a call from a VoIP phone to a regular phone you need a VoIP service provider(eg. Freeway Communications or Vonage) who have equipment that route the calls to the PSTN.

Digital Service Provider

Analog

PSTN

Advantages using VoIP rather PSTN

When you are using PSTN line, you typically pay for time used to a PSTN line manager company: more time you stay at phone and more you'll pay. In addition you couldn't talk with other that one person at a time. In opposite with VoIP mechanism you can talk all the time with every person you want (the needed is that other person is also connected to Internet at the same time), as far as you want. (money independent) and, in addition, you can talk with many people at the same time. talk with many people at the same time. If you're still not persuaded you can consider that, at the same time, you can exchange data with people are you talking with, sending images, graphs and videos.

Signaling and Media

A VoIP call is composed of two parts. 1.Signaling.- The use of signals for controlling communications. The signaling does the call set up, locate users, tear down sessions. 2.Media- After the call has been setup the media transports the voice packets

PROTOCOLS
Examples of protocols are SIP, H323In computing, a protocol is a convention or standard that controls or enables the connection, communications, and data transfer between two computing endpoints. eg TCP/IP, http, ftp. In the field of telecommunications, a communications protocol is the set of standard rules for data representation, signaling, authentication, and error detection required to send information over a channel

Types of VoIP Architecture

PC-to-PC PC-to-Phone Phone-to-Phone Note: In the above listing phones can be either analog or digital phones

Alternative VoIP Architectures

Phone to Internet to Gateway to PSTN

Internet
GATEWAY

PSTN

PC to Phone Connection
Made over the Internet for connecting PC to phones Sample Product: o Net2Phone Need to pay for the calls but they are relatively inexpensive o Cheaper compared to phone to phone calls made over the Internet

Case 2:PC to PC Connection

Made over the internet for voice connection Sample product: o Net2Phone o NetMeeting Calls are free for example, MSN Messenger or Skype.

Internet

Case 3: Phone to Phone Connection

Phone to phone calls are made over the Internet A special phone will connect to a hub or switch on the network PSTN to PSTN when the caller would use a special adaptor for his telephone for example, BT Broadband Voice or Gossiptel.

Internet

Alternative VoIP Architectures

PSTN to Gateway to Internet to Gateway to PSTN

Internet
GATEWAY GATEWAY

PSTN

PSTN

Typical Layout

ATA

VoIP Phone

Cisco 7970 IP Phone

Main regulatory issues around VoIP

Emergency calls Call interception Caller location Numbering Directory assistance & phonebooks Universal service PSTN availability

Advantages of VoIP
Greater Efficiency Higher Reliability Supporting Innovation Cost reduction - low cost phone calls. Convergence of data/voice networks unification. Simplification and consolidation - centralized management.

Customer Demands of Voice Today versus National Security

A multitude of computers, hosts, routers, and other devices, made by multiple vendors Support network growth without regard to end-toend planning. Support network growth without affecting the (public) Internet Support the implementation and deployment of all types of subscriber based applications without affecting the various network interconnected to the Internet. Cont.

Customer Demands of Voice Today versus National Security

Support operational flexibility. Maintain operation regardless of the network health/status of any interconnected networks. Support backward and forward compatibility. This means more than just simple network growth. The ability to maintain communications between network types regardless of the software or hardware versions in place is an enormous benefit to the users of the Internet. Adaptive/Dynamic Routing This is an important capability during these times of national security.

Quality of Service
QoS involves the following Subjective needs: Does the service meet the customers needs? Is the service easy to use?

QoS parameters can be divided into the following categories: Availability Mean Time Between Failure (MTBF) Reliability Delay both perceived and measured Security Bandwidth Information Loss bit error rate; video and audio

Conclusion
As data traffic continues to increase and surpass that of voice traffic, the convergence and integration of these technologies will not only continue to improve, but also will pave the way for a truly unified and seamless means of communication. Implementing VoIP can provide significant benefits and savings to your company.