Internet Telephony

Published on May 2016 | Categories: Documents | Downloads: 46 | Comments: 0 | Views: 265
of 32
Download PDF   Embed   Report

Comments

Content

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 SDIT Dept. Of ISE 2010

IP/ INTERNET TELEPHONY 2 can be transmitted over the Internet. The ATA(analog telephone adaptor). 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.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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY IP Telephony: PBX Replacement:

3

7040 Corporate/Campus External line 7041 7042 7043 PBX

Another campus

8151 8152 8153

PBX 8154

LAN

Internet

LAN

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

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

4

2. DIFFERENERT TYPES OF IP TELEPHONY
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. SDIT Dept. Of ISE 2010

IP/ INTERNET TELEPHONY

5

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

6

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

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

7

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

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

8

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY 9 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 End system Gateway Signaling server Router PSTN Customer-premises equipment, private branch exchange Signaling gateway Service Control Point (SCP), Service Switching Point (SSP) Service Transfer Point (STP)

Comparable addressing concepts in Internet telephony and the PSTN:
Internet Telephony MAC address IP address SIP URL, H.323 alias PSTN Circuit identifier Routing number (E.164) 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. SDIT Dept. Of ISE 2010

IP/ INTERNET TELEPHONY

10

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. SDIT Dept. Of ISE 2010

IP/ INTERNET TELEPHONY H.323 is a standard produced by the ITU-T Study Group 16

11

.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 multipointmultimedia 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-topeer relationships. SIP uses requests and responses to establish, maintain, and terminate calls (or sessions) between two or more end points. SIP-based IP SDIT Dept. Of ISE 2010

IP/ INTERNET TELEPHONY 12 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: u 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

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY Bob signs up for the service from the web as [email protected] He registers from multiple phones Alice tries to reach Bob INVITE ip:[email protected]

13 sipd canonicalizes the destination to

sip:[email protected]

sipd rings both e*phone and sipc Bob accepts the call from sipc and starts talking

SIPD

Call Bob

SIP proxy, redirect server

DB
e*phone Hardware Internet (SIP) phones

sipc

ecse.rpi.edu
Software SIP user agents

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY 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)

14

1

2 3

4 4

5

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

15

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

16

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

17

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 Dept. Of ISE 2010

SDIT

IP/ INTERNET TELEPHONY

18

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

19

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

20

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

21

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY
Gatekeeper cloud 1 2 3 4 5 6 7 8 ARQ ACF/ARJ Setup Setup ARQ ACF/ARJ Connect Connect

22

1

2

3

8

4

5

6

7

Endpoint 1

Endpoint 2

Call Signalling Channel Messages RAS Channel Messages

T1521280-96

Direct Call Signaling: During the admission confirmation, the gatekeeper indicates that the endpoints can exchange call-signaling messages directly.

Gatekeeper cloud

1 1 2 3 4 5 6 ARQ ACF/ARJ Setup ARQ ACF/ARJ Connect

2 3

4

5

Endpoint 1

6

Endpoint 2
T1521290-96

Call Signalling Channel Messages RAS Channel Messages

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

23

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. SDIT Dept. Of ISE 2010

IP/ INTERNET TELEPHONY

24

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 Its software oriented nature will make it to be easily extended and integrated with other Internet telephony with an intranet enables users to save on long-distance bills between terms of time and money than in the traditional PSTN environment. services and applications. sites; they can make point-to-point phone calls via gateway servers attached to the local area network. • 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 Packet switched which allows all of the bandwidth to be used. Circuit switched results in

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY 25 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 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 Auto-attendant. You can have unlimited auto-attendants, one speaking to you while you 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 On-screen consoles and monitoring. Record calls, see who is on the phone, who is talking, 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) Disaster recovery. Soft switch systems (telephone systems that can run on your server) also the Internet providing you with free inter-office calls.

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

provide you key information when a call is received. who is not in the office.

Use a single cable to each desk for both telephone and data. allow you to backup the files and configuration, meaning any problems can often be quickly fixed.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

26

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY •

27

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

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.

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

28

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

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

29

10.CONLUSION
a 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. a 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. a systems a a Internet Telephony is a revolutionary technology that has the potential to completely If you're interested in trying Internet Telephony, then you should check out some of rework the world's phone systems. 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. a 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. a 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. Future: Integration with Web and long-term replacement for current telephone

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY 30 a 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/

SDIT

Dept. Of ISE

2010

IP/ INTERNET TELEPHONY

31

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
SDIT Dept. Of ISE 2010

IP/ INTERNET TELEPHONY

32

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

SDIT

Dept. Of ISE

2010

Sponsor Documents

Or use your account on DocShare.tips

Hide

Forgot your password?

Or register your new account on DocShare.tips

Hide

Lost your password? Please enter your email address. You will receive a link to create a new password.

Back to log-in

Close