INDEPENDENT SEMINAR

ON

BROADBAND

DRONACHARYA COLLEGE OF ENGINEERING

GURGAON

SUBMITTED BY:-

MOHIT ARORA

10191

E.C.E-1(b)

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Table of Contents

Overview of VoIP………………………………………………………………………...4

VoIP Components:

Terminals ............................................................................................................................5

Gateways.............................................................................................................................6

Gatekeepers.........................................................................................................................7

Multipoint control unit……………………………………………………………………7

VoIP Protocols:

H-323 .................................................................................................................................8

Session initiation protocol……………………………………………………………......10

VoIP Signaling and routing …………………………………………………………....12

Benefits and requirements of VoIP….………………………………………………....15

Conclusion………………………………………………………………………………17

References………………………………………………………………………………18

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VOICE OVER INTERNET PROTOCOL

OVERVIEW OF VoIP

Voice over IP is the transport of voice using the Internet Protocol (IP) however this broad term

hides a multitude of deployments and functionality. So voice over Internet Protocol is a method

for taking analog audio signals and turning them into digital data that can be transmitted over the

Internet.

The following types of VoIP applications are in use:

ATA

The simplest and most common way is through the use of a device called an ATA (analog

telephone adaptor). The ATA allows you to connect a standard phone to your computer or your

Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the

analog signal from your traditional phone and converts it into digital data for transmission over

the Internet.

IP Phones

These specialized phones look just like normal phones with a handset, cradle and buttons. But

instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet

connector. IP phones connect directly to your router and have all the hardware and software

necessary right onboard to handle the IP call. Wi-Fi phones allow subscribing callers to make

VoIP calls from any Wi-Fi hot spot.

Computer-to-computer

This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls.

There are several companies offering free or very low-cost software that you can use for this type

of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet

connection.

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VoIP COMPONENTS

TERMINALS

IP Phones

An IP phone uses Voice over IP technologies allowing telephone calls to be made over an IP

network such as the internet instead of the ordinary PSTN system. Calls can traverse the Internet,

or a private IP Network such as that of a company. The phones use control protocols such as

Session Initiation Protocol, Skinny Client Control Protocol or one of various proprietary

protocols such as that used by Skype. IP phones can be simple software-based Soft phones or

purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone.

Ordinary PSTN phones are used as IP phones with analog telephony adapters (ATA). H.323

protocol and provide real-time, two-way multimedia communications. In the case of voice, the

H.323 terminal is generally an IP telephone.

Analog Phones

A telephone can be a basic push-button wall unit or an integrated system complete with

answering machine, stored-number dial, speaker phone, and 900MHz cordless operation.

Computers

With VoIP software such as Skype, yahoo, netmeeting and many more running on computers

they can also be used as communication devices. H.323 is also widely deployed on PCs. A very

common application of the H.323 protocol can be found in the Microsoft NetMeeting software

that allows for both voice and video transmissions on a user’s PC.

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Gateways

Gateways work as a translator to allow communications between H.323 and non-H.323 entities

(for instance, between H.323 terminals and telephones on the circuit-switched network). H.323

gateways provide a means for an H.323 network to communicate to other networks, most

typically the PSTN or PBX systems. In order to provide this interoperability, gateways provide

for translation and call control functions between the two dissimilar network types. Encoding,

protocol, and call control mappings occur in gateways between two endpoints. Gateways provide

many functions, including:

Translating protocols

The gateway acts as an “interpreter,” allowing the PSTN and the H.323 network to talk to

each other to set up and tear down calls.

Signaling Gateway

The Signaling Gateway is located in the service provider’s network and acts as a gateway

between the call agent signaling and the SS7-based PSTN. It can also be used as a signaling

gateway between different packet based carrier domains. It may provide signaling

translation, for example between SIP and SS7 or simply signaling transport conversion e.g.

SS7 over IP to SS7 over TDM.

Trunking Gateway

The Trunking Gateway is located in the service provider’s network and as a gateway between

the carrier IP network and the TDM (Time Division Multiplexing)-based PSTN. It provides

transcoding from the packet based voice, VoIP onto a TDM network. Typically, it is under

the control of the Call Agent / Media Gateway Controller through a device control protocol

such as H.248 (Megaco) or MGCP.

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Access Gateway

The Access Gateway is located in the service provider’s network. It provides support for

POTS phones and typically, it is under the control of the Call Agent / Media Gateway

Controller through a device control protocol such as H.248 (Megaco) or MGCP.

Subscriber Gateway

The Subscriber Gateway is located at the customer premises and terminates the WAN (Wide

Area Network) link (DSL, T1, fixed wireless, cable etc) at the customer premises and

typically provides both voice ports and data connectivity. Usually, it uses a device control

protocol, such as H.248 (Megaco) or MGCP/NCS, under the control of the Call Agent. It

provides similar function to the Access Gateway but typically supports many fewer voice

ports.

Gatekeepers

Gatekeepers provide call control functions such as address translation and bandwidth

management and are often considered to be the most important component in the H.323 stack.

Gatekeepers in H.323 networks are optional. However, if they are present, it is mandatory that

endpoints use their services. The H.323 standards define several mandatory services that the

gatekeeper must provide and specify other optional functionality.

Multipoint Control Units

MCUs provide conference facilities for users who want to conference three or more endpoints

together. MCUs provide a unique function to the H.323 protocol in that they do not provide a

direct interconnection to the H.323 protocol stack. Rather, they provide a method for H.323 to

interconnect voice and videoconferencing. MCUs provide conference support for three or more

endpoints. All terminals participating in the conference establish a connection with the MCU. It

manages conference resources and negotiations between endpoints to determine which audio or

video codec to use.

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VoIP PROTOCOLS

H.323

H.323 is probably the most important standard supporting packetized voice technology. H.323 is

an ITU-T recommendation umbrella set of standards that defines the components, protocols, and

procedures necessary to provide multimedia (audio, video, and data) communications over IP-

based networks. Essentially, H.323 provides a method to enable other H.32X-compliant products

to communicate. In addition to control and call setup standards, H.323 encompasses protocols for

audio, video, and data as follows:

Audio

The compression algorithms H.323 supports for audio are all proven International

Telecommunications Union (ITU) standards (G.711, G.723, and G.729). Because audio is

the minimum service provided by the H.323 standard, all H.323 terminals must have

support for at least one audio codec support, as specified by G.711.

Video

Video capabilities for H.323 are optional. However, any video enabled H.323 terminal

must support the ITU-T H.261 encoding and decoding recommendation.

Data

H.323 references the T.120 specifications for data conferencing. An ITU standard, T.120

addresses point-to-point and multipoint data conferences. It provides interoperability at the

application, network, and transport levels.

The H.323 Protocol Stack

Just as with the TCP/IP protocol, the H.323 protocol is actually a suite of protocols that work

together to provide end-to-end call functionality in a converged network. However, the H.323

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protocol also relies heavily on the services provided by other protocols such as TCP, IP, and

UDP as well as RTP. The protocols that make up the H.323 protocol are Registration,

Admission, and Status (RAS), H.245, and H.225.

Codecs

Coder/decoders (codecs) are used by not only the H.323 protocol but by all VoIP protocols to

define the degree of compression and decompression algorithms that will be used to transport

either a voice or video transmission across a converged network.

Speech codecs, sometimes called voice encoders or vocoders if source codecs are

used, can be divided into three basic classes: waveform, source, and hybrid.

Waveform codec These are older, operationally used high bit rates and provide very good quality speech reproduction.

Source codec

These operate at very low bit rates but tend to produce speech that sounds artificial or tinny.

Hybrid codec These use techniques from both source and waveform coding, operate at intermediate bit rates, and provide good-quality speech.

Fig No 8.1 - The H.323 Protocol Stack

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Session Initiation Protocol

SIP is a simple signaling protocol used for Internet conferencing and telephony. SIP is fully

defined in RFC 2543. Based on the Simple Mail Transport Protocol (SMTP) and the Hypertext

Transfer Protocol (HTTP), SIP was developed within the IETF Multiparty Multimedia Session

Control (MMUSIC) working group. SIP specifies procedures for telephony and multimedia

conferencing over the Internet. SIP is an application-layer protocol independent of the

underlying packet protocol (TCP, UDP, ATM, X.25). SIP is based on a client/server architecture

in which the client initiates the calls and the servers answer the calls. Because it is an open

standard based protocol, SIP is widely supported and is not dependent on a single vendor’s

equipment or implementation. However because of its simplicity, scalability, modularity, and

ease with which it integrates with other applications, this protocol is attractive for use in

packetized voice architectures.

Some of the key features that SIP offers are:

Address resolution, name mapping, and call redirection

Dynamic discovery of endpoint media capabilities by use of the Session Description

Protocol (SDP)

Dynamic discovery of endpoint availability

Session origination and management between host and endpoints

SIP has learned from HTTP and SMTP and has built a rich set of extensibility and

compatibility functions.

SIP was designed to be highly modular. A key feature is its independent use of protocols.

SIP has the capability to integrate with the Web, e-mail, streaming media applications,

and other protocols.

.syngress.comSession Initiation Protocol Components

The SIP system contains two components:

User agents Network servers

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A user agent (UA) is SIP’s endpoint, which makes and receives SIP calls. The client is called the

user agent client (UAC) and is used to initiate SIP requests.

The server is called the user agent server (UAS), receiving the requests from the UAC and

returning responses for the user. There are three kinds of SIP servers:

Proxy server

Proxy servers decide to which server the request should be forwarded and then forward

the request. The request can actually traverse many SIP servers before reaching its

destination. The response then traverses in the reverse order. A proxy server can act as

both a client and server and can issue requests and responses.

Redirect server

Unlike the proxy server, the redirect server does not forward requests to other servers.

Instead, it notifies the calling party of the actual location of destination.

Registrar server

Provides registration services for UACs for their current locations. Registrar servers are

often placed with proxy and redirect servers.

Fig No 8.2 - SIP Components

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VoIP SIGNALING AND ROUTING

In telephony, the signaling information is used to exchange information between endpoints on a

network to set up, control, and end calls. The signaling method that's used depends on the type of

device that's being used and the type of signaling method that's used by the telephone company.

On the PSTN local loop

An open circuit with no current flowing indicates an on-hook condition (telephone

handset placed in the cradle).

Offhook (telephone receiver off the cradle) is indicated by a closed circuit with current

continuously flowing.

DP and DTMF are the address-signaling methods implemented from telephone to switch

in the telephone network.

Earth and magnet (E&M) signaling is the most commonly utilized method of analog

trunking.

VoIP Signaling

In connectionless network architectures such as IP networks, the responsibility for session

establishment and signaling resides in the end stations. To successfully emulate voice services

across an IP network, enhancements to the signaling stacks are required. Some are:

H.323 agent is added to the router for standards-based support of the audio and signaling

streams.

The Q.931 protocol is used for call establishment and tear-down between H.323 agents or

end stations.

Real-Time Control Protocol (RTCP) provides for reliable information transfer once the

audio stream has been established. A reliable session-oriented protocol such as TCP is

deployed between end stations to carry the signaling channels.

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RTP, which is built on top of UDP, is used to transport the real-time audio stream. RTP

uses UDP as a transport mechanism because it has lower delay than TCP and because

actual voice traffic, unlike data traffic or signaling, tolerates low levels of loss and cannot

effectively exploit retransmission.

H.245 control signaling is used to negotiate channel usage and capabilities. H.245

provides for capabilities exchange between endpoints so that codecs and other parameters

related to the call are agreed upon between the endpoints. It is within H.245 that the audio

channel is negotiated.

wyngress.coVoIP signaling is most commonly used in three distinct areas:

signaling from the PBX to the router

signaling between routers

signaling from the router to the PBX.

Signaling Between Routers and PBXs

When signaling from PBX to router, the user picks up the handset, signaling an off-hook

condition. The connection between the PBX and router appears as a trunk line to the PBX, which

signals the router to seize the trunk. Once a trunk is seized, the PBX forwards the dialed digits to

the router in the same manner the digits would be forwarded to a telephone company switch or

another PBX. The signaling interface from the PBX to the router may be any of the common

signaling methods used to seize a trunk line, such as FXS, FXO, E&M, or T1/E1 signaling. The

PBX then forwards the dialed digits to the router in the same manner as the digits would be

forwarded to a telco switch. The PBX seizes a trunk line to the router and forwards the dialed

digits. Within the router, the dial plan mapper maps the dialed digits to an IP address and

initiates a Q.931 call establishment request to the remote peer router that is indicated by an IP

address.

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Fig No 9.1- PBX-to-Router Signaling

Fig No 9.2 - Router-to-Router Signaling

When the remote router receives the Q.931 call request, it signals a line seizure to the PBX. After

the PBX acknowledges this seizure, the router forwards the dialed digits to the PBX and signals

a call acknowledgment to the originating router.

Fig No 9.3 - Router-to-PBX signaling

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BENEFITS AND REQUIREMENTS FOR VoIP

For service providers examining the business case for VoIP, the ubiquity of IP as a networking

technology at the customer premises opens the possibility of deploying a vast range of innovative

converged voice and data services that simply cannot be cost effectively supported over today’s

PSTN infrastructure.

IP-based internet applications, such as email and unified messaging, may be seamlessly

integrated with voice application

IP centrex services allow network operators to provide companies with cost effective

replacements for their ageing PBX infrastructure

VoIP services can be expanded to support multimedia applications, opening up the

possibility of cost effective video conferencing, video streaming, gaming or other multi-

media applications.

The flexibility of next generation platforms allows for the rapid development of new

services and development cycles are typically shorter than for ATM or TDM-based

equipment.

VoIP products based on the MSF architecture, unlike legacy TDM switches, often

support open service creation environments that allow third party developers to invent

and deliver differentiated services.

VoIP leverages data network capacity removing the requirement to operate separate voice

and data networks.

IP equipment is typically faster and cheaper than ATM or TDM-based equipment – a gap

that is increasing rapidly every few months.

Re-routing of IP networks (e.g. with MPLS) is much cheaper than, say, SDH protection

switching.

Whatever the justifications, most service providers recognize that VoIP is the direction of the

future – however when looking at a future PSTN scale solution service providers must ensure

that the following key requirements are met to provide equivalence with the PSTN:

Security

Quality of Service

Reliability

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Migration path

OSS support

Billing

Network Interconnection

These issues are by no means simple and in many cases have delayed roll out of VoIP services.

This white paper will look in more detail at these problems and consider at a high level how they

might be addressed.

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CONCLUSION

Voice over IP is quickly becoming readily available across much of the world, however many

problems still remain. For the time being transmission networks involve too much latency or

drop too many packets, this effects quality of service sometimes severely deteriorating the

quality of the call. Also VOIP contains many security risks, sending out packets that any person

may intercept. Although VOIP may offer cheaper solutions for many the PSTN offers a high

QoS and greater security that makes up for its higher prices. It is my belief that the telephone

market will continue to be dominated by the PSTN until quality of service and security issues

can be addressed.

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REFRENCES

1. www.wikipedia.org 2. www.britanica.com 3. www.computer . howstuffworks.com 4. Data Communications and Networking by Behrouz A. Forouzan

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