ip internet telephony

IP/ INTERNET TELEPHONY 1

1. INTRODUCTION

Today IP Telephony is a very powerful and economical communication options.IP telephony is

the integration and convergence of voice and data networks, services, and applications. Internet

telephony uses the Internet to send audio, video and data between two or more users in the real

time. It is a communications protocol developed to support a packet-switched network. The main

motivation of development of IP Telephony is the cost saving & integrating new services.

Vocaltec introduced the first Internet telephony software product in early 1995, running a

multimedia PC, the Vocaltec Internet Phone. In 1996, Vocaltec announced it was working with an

Intel Company (Dialogic Corporation, an Intel acquisition made in 1999) to produce the first IP

telephony gateway. The technology has improved to that point where conversations are easily

possible. Gateways are the key to bringing IP telephony into the mainstream. By bridging the

traditional circuit-switched telephony world with the Internet.

The basic steps involved in originating the internet telephone calls are conversion of anolog

voice signal to digital format and compression/translation of the signal into internet protocol (IP)

packets for transmission over the internet using ATA(Analog Telephone Adapter ). The process is

reversed at the receiving end.

1.1 What is internet telephony (IP Telephony):

IP Telephony Adds interactive multimedia to the web. Being able to do basic telephony on IP

with a variety of devices.IP telephony is new age technology that banks on Internet connectivity.

The system enables telephone calls empowered by a special IP or Internet protocol network, rather

than the previously patronized PSTN or public switched telephone network.IP Telephony also

works along systems fitted within WiFi enabled mobile phones, PDAs and analog telephony

adapters.

Definition:IP telephony (Internet Protocol telephony) is a general term for the technologies

that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms

of information that have traditionally been carried over the dedicated circuitswitched connections of

the public switched telephone network (PSTN).

Internet Telephony, or Voice over Internet Protocol, is a method for taking analog audio

signals, like the kind you hear when you talk on the phone, and turning them into digital data that

can be transmitted over the Internet. The ATA(analog telephone adaptor). is an analog-to-digital

SDIT Dept. Of ISE 2010


converter. It takes the analog signal from your traditional phone and converts it into digital data for

transmission over the Internet.ATA (analog telephone adaptor). The ATA allows you to connect a

standard phone to your computer or your Internet connection for use with Internet Telephony.

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

1.2 How IP Telephony works:Requirements:The IP telephony uses the broadband internet access for the transmission of voice

over the internet. The basic equipment required, is a broadband connection and a desktop computer

or special IP phones. Computers are convenient in IP telephony, as then, the voice transmission

requires only software and inexpensive earphones.

The IP telephony uses the Internet Protocol for communication through the packet-switched

network of TCP/IP protocol suite. In Internet Protocol, the information that is to be transmitted is

divided into a number of chunks called packets. Each computer communicating has a particular IP

address of its own. When the communication takes place, the packets of information contain the IP

address of the sending and receiving computer. These packets are sent to the gateway computer.

The gateway computer reads the receiver address on each of the packets. The gateway computer

then sends the packets to their respective destination. The packets are sent in any order by the IP

protocol. The Transmission Control Protocol ensures that these packets reach the destination

computer in the correct order. The IP is a connectionless protocol and needs no specific physical

medium for communication.

IP Telephony: PBX Replacement:



1.3Features of IP Telephony:IP telephony modern features such as access to:

Caller ID. ID Calling. Automatic use of network-based directories. Conference calls. Call transfer and hold. Storing user name/number in systems facilitated by different service providers. Weather report analysis. Live news. Voice Messaging Faxing, Fax Services, Fax Broadcast UnPBX Speech Recognition Text to Speech

2. DIFFERENERT TYPES OF IP TELEPHONYSDIT Dept. Of ISE 2010

External line

7043

7040

7041

7042

PBX

Corporate/Campus

InternetLAN

8154

8151

8152

8153

PBX

Another campus

LAN


There are four types IP telephony according to terminal equipment and types of network

PC to PC,

Phone to Phone,

PC to Phone, and

Phone to PC.

2.1: PC-to-PC:

The calling and called parties both have computers that enable them to connect to the

Internet, usually via the network of an Internet service provider (ISP). The two correspondents are

able to establish voice communication. Both users have to be connected to the Internet at that time

and use IP telephony software. In this the caller must know the IP address of the called party. . This

type of IP communication is free of cost and distance is not a limit. The only cost incurred, are the

internet charges.



2.2. Phone-to-phone over IP: The calling and called parties are both subscribers to the public telephony network (fixed or

mobile) and use their telephone set for voice communication in the normal way. However, this way

is least preferred by anybody using IP telephony, as the cost incurred is higher than the other

mentioned ways of IP telephony.

There are two methods for communicating by means of two ordinary telephone sets via an IP

or Internet network.

Use of gateways:

One or more telecommunication players have established gateways that enable the

transmission of voice over an IP network in a way that is transparent to telephone users. It works in

“managed IP network” i.e. a network, which has been dimensioned in such a way as to enable voice

to be carried with an acceptable quality of service.



Use of adapter boxes: A number of companies market boxes, which resemble modems and are installed between the

user's telephone set and his connection to the PSTN.

The calling party initiates his call in the same way as in a conventional telecommunication

network. The first phase of the call is set-up on that network, however, immediately after this the

boxes exchange the information required for the second phase. Data they have exchanged and the

pre-established parameters, establish a connection between each of the two correspondents and

their respective ISP. Once the call has been established, the boxes locally convert the voice signals

into IP packets to be transported over the Internet



2.3. PC-to-Phone:

When the computerized user wishes to call a correspondent on the latter's telephone set, he

must begin by connecting to the Internet in the traditional manner via the network of his ISP. Once

connected, he uses the services of an Internet telephony service provider (ITSP) operating a

gateway, which ensures access to the point that, is closest to the telephone exchange of the called

subscriber. It is this gateway that will handle the calling party's call and all of the signaling relating

to the telephone call at the called party end. This way of communication using the IP telephony,

uses the software at the computer side, but the calls are charged with some minimal fees.

2.4. Phone-to-PC: The calling party is the telephony user and the called party is the PC user. IP telephony

communication in this way requires a special calling card at the telephone end and it can

communicate only with the computer that has a IP telephony software installed. Although a calling

card is required, it is far cheaper than the traditional way of making long-distance calls



3.HOW IP TELEPHONY IS DIFFERENT

Because of the effects of the Internet environment, Internet telephony has a number of

differences from the traditional telephone networks; many of these differences will effect what

sorts of features are possible, how these features are created, and how their interactions are

managed. In general, the new flexibility the Internet gives telephony allows a wide range of

new possibilities; however, this flexibility also introduces new challenges.

The primary technical difference between the Internet and the PSTN is their switching

architectures. The Internet uses dynamic routing (based on non-geographic addressing) versus the

PSTN which uses static switching (based on geographic telephone numbering). Internet's

"intelligence" is very much decentralized, or distributed, versus the PSTN which bundles transport

and applications resulting in the medium's intelligence residing at central points in the network.

PSTN is circuit switched network. The basic fundamental problem with the past and

existing public telephone network is its reliance on circuit based switching technology.Circuit

switching is a technology that has been used by telephone networks for over 100 years. With circuit

switching, when a telephone call is occurs between two parties, the connection is must be

completely maintained for the entire duration of the call. Because the connection is both directions

(full duplex), this connection is called a circuit. This is the basic process used by the Public

Switched Telephone Network (PSTN It dedicates a fixed amount of bandwidth for each

conversation and thus quality is guaranteed. When the caller places a typical voice call, she picks

up the phone and hears the dial tone. Then she dials the country code, area code, and the number of

callee. The central office will establish the connection, and then the caller and callee can discuss

with each other.

IP Data networks don’t use circuit switching. Instead, data networks use a method called

packetswitching.While circuit switching keeps the connection open and constant the entire time,

packet switching only opens the connection long enough to send a small piece of data, called a

packet, from one computer to the other. The sending computer puts the data into small packets,

with an address on each one telling the data network where to send them. When the receiving

computer gets the packets, it reassembles them into the original data.



Packet switching is a much more efficient way to transmit. It minimizes the time that the

connection must be maintained, which reduces the demand on the network. It also frees up the

computers that are communicating with each other to accept information from other computers at

the same time as well. When the caller places IP telephony call, she picks up the phone and hears

dial tone from the PBX (private branch exchange) if one is available. Then she dials a number

which is forwarded to the nearest IP telephony gateway located between the PBX and a TCP/IP

network . The IP telephony gateway finds a route through the Internet that reaches the called

number. Then the call is established. The IP telephony gateway modulates voice into IP packets and

sends them on their way over the TCP/IP network as if they were typical data packets. Upon

receiving the IP encoded voice packets, the remote IP telephony gateway reassembles them into

analog signals to the callee through the PBX.

Comparable components of Internet telephony and the PSTN:Internet Telephony PSTNEnd system Customer-premises equipment, private branch exchangeGateway Signaling gatewaySignaling server Service Control Point (SCP), Service Switching Point (SSP)Router Service Transfer Point (STP)

Comparable addressing concepts in Internet telephony and the PSTN:Internet Telephony PSTNMAC address Circuit identifierIP address Routing number (E.164)SIP URL, H.323 alias Telephone number, including 800/900 numbers

VoIP technology: uses this same packet-switching technology. For example, packet

switching allows multiple telephone calls to use the same amount of space usually occupied by only

one in a circuit switched network. Using PSTN, that 15 minute telephone call used 15 full minutes

of transmission time at a bandwidth cost of 128 Kbps. With VoIP, that same call may have

occupied only 3.5 minutes of transmission time. Based on this example, three or four additional

calls could have easily fit into the space used by a single call under the circuit switched system. An

additionally, data compression technology could further reduces the size requirements of each call.

. VoIP services carry no other taxing while a traditional phone service billing includes numerous

taxes and other charges.while normal telephones are permanently linked to the telephone lines,

ATAs can be taken anywhere in the world along with you. Then, by attaching it to a normal

telephone and an internet connection, you can make VoIP calls to any other ATA in the same

network for no additional cost.



4.DIFFERENT TYPES OF PROTOCALS:

Different type of standard and protocols are employed by the IP telephony.

• H.323 Protocol, • Media Gateway Control Protocol (MGCP), • Skinny Client Control Protocol (SCCP), • Session Initiation Protocol (SIP),

4.1 H.323:

In order for the internet to provide useful services, Internet telephony required a set of control

protocols for connection establishment, capabilities exchange as well as conference control. This

was the basis for H.323. H.323 provides the call set up and signaling functionality’s as well as

providing the gateway, which makes interoperation of different networks possible. IP telephony

Systems incorporate these protocols in their functionality’s to ensure better Quality of Service and

the smooth transfer of packets over the Internet Protocol, which was designed to mainly transport

data packets.

H.323 is a standard that specifies the components, protocols and procedures that provide

multimedia communication services such as real-time audio, video, and data communications over

packet networks, including Internet Protocol (IP) based networks.



H.323 is a standard produced by the ITU-T Study Group 16

.H.323 is part of a family of ITU-T recommendations called H.32x that provides multimedia

communication services over a variety of networks. H.323 can also be applied to multipoint-

multimedia communications. Currently the most widely supported IP telephony signaling protocol.

One of the primary goals in the development of the H.323 standard was the interoperability

with other multimedia-services networks. This interoperability is achieved through the use of a

gateway. A gateway performs any network or signaling translation required for interoperability

4.2 MGCP:

MGCP provides powerful, flexible and scalable resource for call

control.Cisco Unified CallManager uses MGCP to control media on the

telephony interfaces of a remote gateway and also uses MGCP to deliver

messages from a remote gateway to appropriate devices.

MGCP enables a call agent (media gateway controller) to remotely control

and manage voice and data communication devices at the edge of multiservice

IP packet networks. Because of its centralized architecture, MGCP simplifies the

configuration and administration of voice gateways and supports multiple

(redundant) call agents in a network. MGCP does not provide security

mechanisms such as message encryption or authentication.

The MGCP gateway provides call preservation (the gateway maintains

calls during failover and fallback), redundancy, dial-plan simplification (the

gateway requires no dial-peer configuration), hook flash transfer, and tone on

hold. MGCP-controlled gateways do not require a media termination point

(MTP) to enable supplementary services such as hold, transfer, call pickup, and

call park. If the MGCP gateway loses contact with its Cisco Unified CallManager,

it falls back to using H.323 control to support basic call handling of FXS, FXO,

T1 CAS, and T1/E1 PRI interfaces.

4.3 SIP :

ASCII-based SIP works in client/server relationships as well as in peer-to-

peer relationships. SIP uses requests and responses to establish, maintain, and

terminate calls (or sessions) between two or more end points. SIP-based IP



telephony using a pure P2P architecture instead of static set of SIP servers improves the reliability

and allows the system to dynamically adapt to node failures.

4.3.1. SIP Functionality: IETF-standardized peer-to-peer signaling protocol (RFC 2543): Locate user given email-style address Setup session (call) (Re)-negotiate call parameters Manual and automatic forwarding Personal mobility: different terminal, same identifier Call center: reach first (load distribution) or reach all (department conference) Terminate and transfer calls Lightweight multimedia session initiation, call control, capabilities exchange, and user location Based on http; textual, reuses authentication mechanisms. Provides full telephony services: call forward, transfer, 800,900 style numbers Uses SDP (Session Description Protocol) for expressing capabilities

4.3.2 Basic methods in SIP:

INVITE - ask a user to join a session; callee responds with accept or reject, along with a slew of reason codes

OPTIONS - obtain capabilities, but don’t invite CONNECTED - acknowledges acceptance BYE - for transfers and session terminations REGISTER - Allows a user to register with a SIP server

4.3.3 SIP-based telephony Call: Session Initiation Protocol – SIP

o Contact “office.com” asking for “bob”

o Locate Bob’s current phone and ring

o Bob picks up the ringing phone



SIP-based telephony has client-server architecture. As shown in Fig. 2, when a user, Bob,

starts the SIP client on his PC, IP-phone or hand-held device, the client registers with the SIP server

indicating the IP address of the device. The SIP server stores the mapping between the identifier

[email protected] and the IP address.

When another user, Alice, makes a call or sends instant message for [email protected] to the

server in home.com domain,the server proxies the request to the current device of Bob.


SIP proxy,redirectserver

DB

SIPD

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

Call Bob

Bob signs up for the service from the web as [email protected] registers from multiple phonesAlice tries to reach Bob INVITE ip:[email protected]

sipd canonicalizes the destination to sip:[email protected] rings both e*phone and sipcBob accepts the call from sipc and starts talking

ecse.rpi.edu

mailto:[email protected]


4.3.4 IP SIP Phones and Adaptors: Are true Internet hosts:

• Choice of application• Choice of server• IP appliances

Implementations:• 3Com (3)• Columbia University• MIC WorldCom (1) • Mediatrix (1)• Nortel (4)• Siemens (5)


4

1

3

2

54


5. H.323 COMPONENTS

The H.323 standard specifies four kinds of components, which, when networked together,

provide the point-to-point and point-to-multipoint multimedia communication Services:

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.



5.1. Terminals: H.323 terminal can either be a personal computer (PC) or a stand-alone device, running

an H.323 and the multimedia applications. It supports audio communications and can optionally

support video or data communications. Because the basic service provided by an H.323 terminal is

audio communications, The primary goal of H.323 is to inter work with other multimedia

terminals.H.323 terminal plays a key role in IP-telephony services.

5.2. Gateways:

A gateway connects two dissimilar networks. An H.323 gateway provides connectivity

between an H.323 network and a non-H.323 network. A gateway is not required, however, for

communication between two terminals on an H.323 network.



For example an H.323 gateway can provide connectivity between a circuit switched

network, such as the PSTN and an H.323 terminal. The connectivity of these dissimilar networks

however has to be achieved by using translation protocols for call set up and release, and

transferring information between the networks connected by the gateway. A gateway is although

not required for communicating between two terminals on an H.323 network.

The way the gateway works is that on the H.323 side a gateway runs H.245 control

signaling for exchanging capabilities, H.225 call signaling for call set-up and release, and H.225

registration, admissions and status (RAS), for registration with the gatekeeper. On the SCN side the

gateway runs SCN specific protocols such as ISDN and SS7 protocols.

• Gateway– Interface between H.323 systems and other systems - PSTN, H.324 (PSTN

multimedia), H.320 (ISDN multimedia), H.321 (ATM multimedia)

5.3 Gatekeepers: A gatekeeper can be considered to be the controller of an H.323 network. It

provides call control services such as address translation and bandwidth management as

defined within RAS.

• Gatekeeper– Controls sessions– Performs user location and registration– Performs admission control– Reroutes signaling– Processes RAS (Registration, Admissions, Status) from H.323 terminals



A gatekeeper has many functions: 1. Address Translation:

The gatekeeper translates this E.164 telephone number or the alias into the network address

for the destination terminal. The destination endpoint can be reached using the network address on

the H.323 network.

2. Admission Control:

The gatekeeper can control the admission of the endpoints into the H.323 Network by using

RAS messages, admission request (ARQ), confirm (ACF), and reject (ARJ).

3. Bandwidth Control:

The gatekeeper provides support for bandwidth control by using the RAS messages,

bandwidth request (BRQ), confirm (BCF), and reject (BRJ). If a network manager has specified a

threshold for the number of simultaneous connections on the H.323 network, the gatekeeper can

refuse to make any more connections once the threshold is reached.

4. Zone Management:

The gatekeeper provides the above functions address translation, admissions control, and

bandwidth control4or terminals, gateways, and MCUs located within its zone of control

5. Call-Control Signaling: The gatekeeper can route call-signaling messages between H.323 endpoints using H.225

call signaling messages.

6. Call Authorization: Gatekeeper authorizes the user to setup connection within its zone. 7. Call Management: The gatekeeper may maintain information about all active H.323 calls. It can control its zone by providing the maintained information.



5.4Multipoint Control Units:

MCUs provide support for conferences of three or more H.323 terminals. All terminals

participating in the conference establish a connection with the MCU. The gatekeepers, gateways,

and MCUs are logically separate components of the H.323 standard but can be implemented as a

single physical device.



6. PROTOCAL SPECIFIED BY H.323

The protocols specified by H.323 are listed below: 1. Audio CODECs 2. Video CODECs 3. H.225 registration, admission, and status (RAS) 4. H .225 call signaling 5. H.245 control signaling 6. Real-time transfer protocol (RTP) 7. Real-time control protocol (RTCP)

6.1 Audio CODEC: An audio CODEC encodes the audio signal from the microphone for transmission on the

transmitting H.323 terminal and decodes the received audio code that is sent to the speaker on the

receiving H.323 terminal. Audio is the minimum service provided by the H.323 standard, all H.323

terminals must have at least one audio CODEC support. ITU-T G.711 (audio coding at 64 kbps),

G.722 (64, 56, and 48 kbps), G.723.1 (5.3 and 6.3 kbps), G.728 (16 kbps), and G.729 (8 kbps)

recommendation are the audio CODEC.



6.2 Video CODEC:

A video CODEC encodes video from the camera for transmission on the transmitting

H.323 terminal and decodes the received video code that is sent to the video display on the

receiving H.323 terminal. The support of video CODECs is optional. ITU-T H.261 is the video

CODEC recommendation.

6.3 H.225 Registrations, Admission, and Status (RAS): The RAS channel is a User Datagram Protocol (UDP) - based protocol.RAS is the

protocol between endpoints (terminals and gateways) and gatekeepers.

RAS is used to perform these tasks:

Gatekeeper discovery (GRQ):

Endpoint registration/deregistration

Endpoint location

Admission control

Gatekeeper Discovery: The gatekeeper discovery process is used by the H.323 endpoints to

determine the gatekeeper with which the endpoint must register.

Endpoint Registration: Registration is a process used by the endpoints to join a zone and inform

the gatekeeper of the zone's transport and alias addresses.

Endpoint Location: Endpoint location is a process by which the transport address of an endpoint is

determined and given its alias name or E.164 address.

Admission Control: The gatekeeper can control the admission of the endpoints into the H.323

network. It uses RAS messages, admission request (ARQ), confirm (ACF), and reject (ARJ).

6.4 H.225 Call Signaling: The H.225 call signaling is used to establish a connection between two H.323 endpoints over

which the real-time data can be transported. There are the two type of Call Signaling.

Gatekeeper-Routed Call Signaling:

The gatekeeper receives the call-signaling messages on the call signaling channel from one

endpoint and routes them to the other endpoint on the call-signaling channel of the other endpoint.



Direct Call Signaling: During the admission confirmation, the gatekeeper indicates that the

endpoints can exchange call-signaling messages directly.

6.5 H.245 Control Signaling: H.245 control signaling consists of the exchange of end-to-end H.245 messages between

communicating H.323 endpoints. The H.245 control channel is the logical channel 0 and is

permanently open.

Capabilities Exchange: Capabilities exchange is a process using the communicating terminals’

exchange messages to provide their transmit and receive capabilities to the peer endpoint.

Logical Channel Signaling: A logical channel carries information from one endpoint to another endpoint (in the case of a point-to-point conference) or multiple endpoints.



6.6 Real-Time Transport Protocol: Real-time transport protocol (RTP) provides end-to-end delivery services of real time audio

and video over packet switched networks.. It is used by both SIP and H.323. The transport protocol

must allow the receiver to detect any losses in packets and also provide timing information.

The functions provided by RTP include: Real time Transport Protocol - RTP

Send and receive audio packets

RTP provides for:

Sequencing: The sequence number in the RTP packet is used for detecting lost packets Payload Identification: In the Internet, it is often required to change the encoding of the

media dynamically to adjust to changing bandwidth. Frame Indication: Video and audio are sent in logical units called frames. To indicates the

beginning and end of the frame, a frame marker bit has been provided. Source Identification: In a multicast session, we have many participants. So an identifier is

required to determine the originator of the frame. For this Synchronization Source (SSRC) identifier has been provided.

Intramedia Synchronization: To compensate for the different delay jitter for packets within the same stream, RTP provides timestamps, which are needed by the play-out buffers.

6.7 Real-Time Transport Control Protocol: Real-time transport control protocol (RTCP) is the counterpart of RTP that provides

control services. The primary function of RTCP is to provide feedback on the quality of the data

distribution. In a RTP session, participants periodically send RTCP packets to obtain useful

information about Quos etc.

The additional services that RTCP provides to the participants are:• QoS feedback: RTCP is used to report the quality of service. The information provided includes

number of lost packets, Round Trip Time, jitter and this information is used by the sources to adjust

their data rate.

• Session Control: By the use of the BYE packet, RTCP allows participants to indicate that they

are leaving a session.

• Identification: Information such as email address, name and phone number are included in the

RTCP packets so that all the users can know the identities of the other users for that session.

• Intermedia Synchronization: Even though video and audio are normally sent over different

streams, we need to synchronize them at the receiver so that they play together. RTCP provides the

information that is required for synchronizing the streams.



7. ADVANTAGES OF IP TELEPHONY

Cost: The cost of the IP telephony is very less, as it uses the existing network for

communication. No new communication networks are required to be established for this

technology. Also, the cost for setting up the packet switched networks is very less as compared to

cost of the other networks. The cost further decreases, since the same network is used for

transmitting voice and data. Many services like call forwarding, web conferencing and video calls,

thus, are cheaper using the IP telephony.

Easy management: No need to get a specialist telephony expert out to add a new user, no

need to adjust the wiring to patch in a new phone. Use a simple user interface to enter the name,

and the extension and it's working.

Mobility: The IP telephony is very convenient way of communication. Once you have an

internet connection and your computer, you can make a voice call anywhere in the world.IP

telephony is one of the widely used methods for making regular long distance calls in big

organizations.

Deployment of new Internet telephony: services require significantly lower investment in

terms of time and money than in the traditional PSTN environment.

Its software oriented nature will make it to be easily extended and integrated with other

services and applications.

Internet telephony with an intranet enables users to save on long-distance bills between

sites; they can make point-to-point phone calls via gateway servers attached to the local area

network.

Packet switched which allows all of the bandwidth to be used. Circuit switched results in

gaps.

Scalable: The software oriented nature of IP telephony makes it easily scalable, making it

possible to integrate other services and applications as well. Adding a new phone to the already

running system of IP telephony doesn't require an additional new line.

Improved Voice Quality: with the recent developments, these problems have now been

largely eliminated. Nowadays, IP telephony's voice quality is competitive with that offered by its

rival PSTN and conventional phone systems The delay in transmission has been reduced to an

acceptable 250 milliseconds or less.



Reduced Initial Investment: The installation of new Internet telephony services requires

significantly lower investment as compared to traditional PSTN environment. These savings are

both in terms of time and money, with the addition of new IP phones to the existing system being

cheaper and easier.

Free inter-office calls (multiple sites). If you have many offices, these can be linked over

the Internet providing you with free inter-office calls.

On-screen dialing. Click on an outlook contact and the number is automatically dialed.

Voicemail. You can have unlimited voicemail boxes, and have various rules which determine

which message is played to which caller.

Auto-attendant. You can have unlimited auto-attendants, one speaking to you while you

pick up your messages, another one to a user while they make a choice which extension they wish.

Screen popping. See who is calling you when they call.

Home working. Plug in a phone at home, and automatically receive you work bound calls.

Integration with Microsoft Outlook. Use your contacts, no need to maintain multiple phone lists.

Scripting. Handle a call any way you like.

Database integration. Already have a customer database? Get your internal applications to

provide you key information when a call is received.

On-screen consoles and monitoring. Record calls, see who is on the phone, who is talking,

who is not in the office.

Flexible console. Make you on-screen phone look however you like.

No need for separate cabling. It all runs over your normal network cabling (Cat5)

Use a single cable to each desk for both telephone and data.

Disaster recovery. Soft switch systems (telephone systems that can run on your server) also

allow you to backup the files and configuration, meaning any problems can often be quickly fixed.



8.DISADVANTAGES

The disadvantages of IP services:

Limited or no use in the absence of a dedicated internet access. The system does not

accommodate calls beyond the Local area network or LAN, unless there is an integrated,

compatible PBX system in place.

Total dependency on separate electric connectivity. Unlike the PSTN phones, IP Phones

and routers connect only via mains electricity. The system is not empowered to work via

power generated from telephone exchanges.

Easy congestion. These networks, particularly residential internet connectivity, easily

succumb to congestion. The result is poor voice quality or a complete call-drop, in the midst

of an emergency.

Lapse in connectivity when exposed to high-latency connectivity. The technology does not

empower internet-call connectivity when exposed to latency induced by protocol overhead.

The system also fails to function effectively when exposed to satellite internet integration.

Failure when integrated alongside other digital equipment. The IP telephony technology

becomes redundant when other digital systems are integrated into the adopted phone line.

Equipment like digital video recorders and home security systems do not integrate with VoIP.

Challenging emergency calls. The technology becomes a challenge to surpass an

emergency. VoIP uses special IP-addressed phone numbers and not regular public-service

NANP phone numbers. Hence, it becomes difficult for a 911 operator to identify the exact

geographic location of the given IP address.

Distorted facilitation when challenged by latency and packet-loss. The Internet

connection used by the IP telephony technology makes the integrated system susceptible to

broadband latency, jitter and packet-loss. The result is distorted and garbled communication

due to transmission error.



Exposure to Denial of Service attacks (DoS attack). IP telephony, like other internet

integrated networks, is subjected to Denial of Service break-downs if the address used is an

public IP id.

IP telephony is susceptible to attacks from viruses and hacking. The system uses a

technology that is completely dependent on the power of integrated PCs. The resultant

processor drain leads to frequent communication-quality loss and system-crash. As any other

information stored on your computer and transmitted through Internet Protocol VoIP is

susceptible to viruses and hacking

Much depends on the processor your computer uses and other requirements. If you run

several programs simultaneously your VoIP phone call may be distorted. The program may

either slow down or even crash in the middle of an important conversation.



9.SECURITY ISSUES

The Internet is an open network where everyone can receive and transmit packets

relatively easily. Eavesdropping of calls in IP networks is probably easier than in PSTN. Therefore,

some mechanisms are necessary to avoid eavesdropping. In addition to voice stream also signaling

(call setup, call management, billing) requires protection to prevent spoofing of calls, denial of

service, spamming (disturbing), etc.

IP telephony maintenance primarily involves the following tasks:

• Keeping up-to-date with operating system and third-party service packs to eliminate well-known

security holes .Unfortunately, viruses, worms, and denial-of-service attacks are a part of computer

daily life. We hear of, and experience, the proliferation of Smurf, Code Red, Nimbda, SQL

Slammer, Blaster, Nachi, Sobig, or the virus-de-jour far too often. Anti-virus and intrusion

detection systems go a long way to protect us against the atrocities of these attacks. But the best

way to mitigate these attacks is to keep the operating system up to date.

• Implementing critical support patches on servers and Cisco® devices when appropriate

• Subscribing to mailing lists that publicize urgent vulnerabilities and critical patches

• Updating anti-virus definitions to protect against well-known worms and viruses

• Performing daily backups of servers with periodic data recovery tests



10.CONLUSION

Internet telephony is a powerful and economical communication option that integrates both

telephone networks and data networks together. The ability to use IP networks to carry

traditional telephone traffic brings both challenges and opportunities to all the long-distance

telephone service companies. Although a lot of difficulties exist, from the technological point

of view to social issues, it is believed that it will bring a great change to communication field

and bring a new huge market.

This paper identifies two major primary sources that cause latency in the Internet telephony,

and present means of managing the latency to maintain sufficient quality of service in Internet

telephony.

Future: Integration with Web and long-term replacement for current telephone systems

Internet Telephony is a revolutionary technology that has the potential to completely rework

the world's phone systems.

If you're interested in trying Internet Telephony, then you should check out some of the free

Internet Telephony software available on the Internet. You should be able to download and set

it up in about three to five minutes. Get a friend to download the software, too, and you can

start tinkering with Internet Telephony to get a feel for how it works.

Internet telephony is much cheaper than traditional long distance services. Keeping this rising

demand of Internet telephony in mind, many software companies have entered this field in

order to take a share of this growing market segment.

The availability of free Internet telephony software has helped in increasing the market for

Internet telephony services, which is evident from the number of new users that have

subscribed to these services in recent years. This holds good for the future of Internet

telephony services.

Although it is going take some time to happen, eventually all circuit switched networks will

be replaced with packet switching technology. IP telephony makes sense in economic terms

and infrastructure requirements. More and more businesses are installing VoIP systems, and

the technology will continues to grow in popularity.



11. REFERENCE

[1] Book on “IP Telephony” Olivier Hersent, David Gurle & Jean-Pierre Petit.

[2]

www.iec.org/online /tutorials/

[3] www.cis.ohio-state.edu/~jain/cis788-97/internet_telephony/index.htm

[4] www.cs.columbia.edu/~coms6181/

[5] www.terena.nl/library/ IPTELEPHONYCOOKBOOK/chapters/Chapter4.pdf

[6] www.cisco.com/

[7] www.tmcnet.com/

[8] www.javvin.com/

[9] www.ieee.org/


http://www.iec.org/online


12. GLOSSARY

ARP - Address Resolution Protocol. Internet protocol used to map an IP address to a MAC

address.

ATA - Analog Telephone Adapter

Codec- Coder-decoder.

PBX(Private Branch Exchange) --Digital or analog telephone switchboard located on the

subscriber premises, typically with an attendant console, and used to connect private and public

telephone networks.

PSTN-Public Switched Telephone Network. It refers to the world's collection of interconnected

voice-oriented public telephone networks both commercial and government owned. It is also

referred to as the Plain Old Telephone Service (POTS).

Real-Time Transport Protocol (RTP) -RTP is designed to provide end-to-end network

transport functions for applications transmitting real-time data, such as audio, video, or simulation

data, over multicast or unicast network services.

Router- An interface device between two networks that selects the best route even if there are

several networks between the originating network and the destination.

SIP--Session Initiation Protocol

Voice over Internet Protocol (VoIP)- Technology used to transmit voice conversations

over a data network using the Internet Protocol (IP).

GSTN: general switched telephone network

CSN: circuit-switched network

SCN: switched circuit network (this is what we’ll use, mostly)


ip internet telephony

Documents

ip telephony gateway

features of ip telephony

introductiontoday ip

internet protocol ip

ip telephony modern

types ip telephony accordin

basic telephony

telephony world