internet telephony - report

8/8/2019 Internet Telephony - Report

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A Seminar

On

Internet Telephony

Presented by

Nitin Prakash Sharma

M.Tech. IT 1st year

Indian Institute of Technology, Kharagpur

Under the guidance ofDr. S.K. Ghosh

Assistant Professor

School of Information Technology

Indian Institute of Technology, Kharagpur


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Contents

Abstract............................................................................................................................... 1

1. Introduction............................................................................................................... 12. What is internet telephony (IP Telephony)? .......................................................... 12.1. The factors making Internet Telephony possible........................................... 2

3. H.323 Standards for internet telephony.................................................................. 2

3.1. RAS signalling ................................................................................................... 2

3.2. Q.931 signalling ................................................................................................. 3

3.3. H.245 signalling ................................................................................................. 3

3.4. RTP and RTCP (Real-time Transport Protocol and Real-time ControlProtocol)......................................................................................................................... 3

3.5. Resource Reservation Protocol (RSVP).......................................................... 4

3.6. H.323 Signaling Protocol .................................................................................. 5

3.6.1. H.323 Terminals........................................................................................ 63.6.2. H.323 Gateways......................................................................................... 6

3.6.3. Gatekeepers ............................................................................................... 7

3.6.4. Multipoint Control Units (MCUs) ........................................................... 8

4. Classes of connections............................................................................................... 9

4.1. Phone-to-Phone via the Internet...................................................................... 9

4.2. Pone-to-PC in the Internet ............................................................................. 10

4.3. PC in the Internet-to-Phone ........................................................................... 11

4.4. PC-to-PC in the Internet ................................................................................ 12

4.5. PC in the Internet-to-PC in the separate IP-based network via SCN ........ 13

4.6. PC in the separate IP-based network-to-PC in the Internet via SCN........ 14

4.7. PC in the separate IP-based network-to-PC in the separate IP-basednetwork via SCN ......................................................................................................... 16

5. Requirements for Internet Telephony Management ........................................... 17

5.1. Pre-Deployment Assessment .......................................................................... 17

5.2. Post-Deployment QoS Monitoring ................................................................ 18

5.3. Performance from the End-User Perspective............................................... 18

5.4. Managing Security .......................................................................................... 18

5.5. Troubleshooting .............................................................................................. 18

5.6. Automated Management ................................................................................ 18

5.7. Call Management ............................................................................................ 19

6. Conclusion ............................................................................................................... 19

7. References ................................................................................................................ 19


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Internet Telephony 1/19

AbstractCommunication via packet and data networks such as IP, ATM, Frame Relay has

become a preferred strategy for both corporate and public networks. Experts predict that

data traffic will soon exceed telephone traffic if it already hasnt. As a result of this there

has been considerable interest in transmitting traditional telephone traffic over data

networks. Internet Telephony is a powerful and economical communication options. It isa 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 circuit-switched connections of the public switched

telephone network (PSTN). It is based on IP networking, which offers the potential for

much more than just telephony. ). The seminar will attempt to provide a basic

understanding of what Internet telephony is and some of the protocols used in it. It also

covers type of connections and addressing used for those connections in Internet

telephony, with a brief description of requirements for Internet telephony management.

1. IntroductionIP telephony uses the Internet to send audio, video, fax etc between two or more

users in real time, so the users can converse. VocalTec* introduced the first IP telephony

software product in early 1995. Running a multimedia PC, the VocalTec Internet Phone*(and the numerous similar products introduced since) lets users speak into their

microphone and listen via their speakers.

Within a year of its birth, IP telephony technology had caught the world's attention.The technology has improved to a point where conversations are easily possible. And it

continues to get better. Dozens of companies have introduced products to commercialize

the technology, and virtually every major telecommunications company has launchedresearch to better understand this latest threat to its markets.

In March of 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 original Internet telephone products based on multimedia PCsare tremendous - offering the ability to combine voice and data on one network. They

also offer low-cost long distance "telephone" service (assuming the user already has a

multimedia PC and a fixed-rate Internet service provider [ISP] account).Gateways are the key to bringing IP telephony into the mainstream. By bridging the

traditional circuit-switched telephony world with the Internet, gateways offer the

advantages of IP telephony to the most common, cheapest, most mobile, and easiest-to-use terminal in the world: the standard telephone. Gateways also overcome another

significant IP telephony problem: addressing. To address a remote user on a multimedia

PC, you must know the user's Internet Protocol (IP) address. To address a remote user

with a gateway product, you only need to know the user's phone number.

2. What is internet telephony (IP Telephony)?When the concept of IP telephony first emerged, it represented a revolution in the

way long distance telephone calls could be conducted. Today, however, IP telephony


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embodies much more than cheaper long distance calls for friends and families. Bytextbook 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 circuit-

switched connections of the public switched telephone network (PSTN).Here it does not matters whether traditional telephony devices, multimedia PCs ordedicated terminals take part in the calls or the calls are entirely or only partially

transmitted over the Internet. Using the Internet or a corporate local or wide area

network, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN.

The challenge in IP telephony is to deliver the voice, fax, or video packets in adependable flow to the user. While most consider IP telephony to be the movement of

real-time voice over IP (VoIP), IP telephony actually embodies much more than that. IP

telephony also delivers application value in non-real time, packet-switchedcommunication namely the transport of voice and fax messages.

2.1.The factors making Internet Telephony possible Voice quality is increasing, thanks to new codec technology There are ongoing improvements in compression techniques Full-duplex PC sound cards enable two-way simultaneous calls The typical PC is getting more and more powerful, making it possible to

perform processor-intensive functions without specialized hardware.

3. H.323 Standards for internet telephonyThe real-time transport protocol along with the real-time transport control protocol

is used to transport real-time data as well as providing QoS feedback. Since IP does not

guarantee Quality of Service the resource reservation protocol is used to reserve

resources such as bandwidth for the duration of a call there by increasing the reliability.In order for the internet to provide useful services, Internet telephony required a set of

control protocols for connection establishment, capabilities exchange as well asconference control. This was the basis for H.323. H.323 provides the call set up and

signaling functionalitys as well as providing the gateway, which makes interoperation of

different networks possible. IP telephony Systems incorporate these protocols in theirfunctionalitys to ensure better Quality of Service and the smooth transfer of packets over

the Internet Protocol, which was designed to mainly transport data packets.

3.1.RAS signallingThe RAS channel is a User Datagram Protocol (UDP)- based protocol that is used

for Endpoint Registration/ Deregistration, Admission Control, Bandwidth ChangeRequest, and Endpoint Status Control. An endpoint can broadcast a request for aGatekeeper with a Gatekeeper Request (GRQ) message (or the Gatekeeper can be

manually configured). Before the endpoints are allowed to make any call, they must

register themselves at the Gatekeeper with a Register Request (RRQ) message. Before

call setup can be initiated, an Access Request Confirm (ARQ) message must besubmitted, stating the called endpoint address and requested bandwidth.


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The RAS channel provides the means to control user access to the network andusage of the network.

3.2.Q.931 signallingThe Q.931 channel is a Transmission Control Protocol (TCP)-based call control

protocol that is used for call setup and call release. The protocol is based on Integrated

Services Digital Network (ISDN) Q.931, which is a well-proven protocol for this type ofconnection-oriented communication. It provides capabilities for handling a variety ofsupplementary services related to specific connections or users and enables interworking

with the SCN.

3.3.H.245 signallingThe H.245 Control channel is a TCP-based protocol that is used for media channel

signalling, handling the channel setup and release, and signalling bandwidth usage for the

media channels. While it is an end-to-end control channel, it can be monitored by theGatekeeper and information such as codec choice and requested bandwidth can be read

from the messages and restricted when necessary. Requests for more bandwidth than is

already reserved for the call (via RAS signalling) can be intercepted and restricted. H.245

has four messages that include request and response messages, enabling the most flexiblebidirectional negotiation. These messages provide the means to negotiate different media

formats in each direction, and they can include several media channels in each directionper call. H.245 is also used to carry Dual Tone Multi Frequency (DTMF) tones end-to-

end.

The H.323 series originally was designed for a LAN environment in which

signalling delay was of little concern. In H.323 Version 2, the scope has changed toencompass packet-based networks in general, which also include WANs. The WAN

change, or the Fast Connect procedure, was introduced to minimise the call setup time.

This method includes the H.245 capability parameter in the setup message and assumesthat capability negotiation is not needed. H.323 Version 2 also includes handling of

supplementary services in the H.450 series, such as Call Transfer, Call Diversion, CallWaiting, etc. These services are handled via the call signalling channel, which conveysthe supplementary service-related information in the user-to-user information element for

a number of message types (Alerting, Call Proceeding, Connect, Setup, Release

Complete, Facility, Progress). For a call-related service invocation, this must be done onthe established call-signalling channel for that call. For a non-call-related service,

invocation of a H.225 call-independent signalling connection is established. This means

that supplementary services can be handled either in conjunction with an actual call or

completely independent of a call. In both cases, the procedure allows for Gatekeepercontrol and billing of service invocations because the H.225 addressing and routing

mechanism is utilised.

3.4.RTP and RTCP (Real-time Transport Protocol and Real-timeControl Protocol)

RTP supports the transfer of real-time media (audio and video) over packet

switched networks. It is used by both SIP and H.323. The transport protocol must allowthe receiver to detect any losses in packets and also provide timing information so that

the receiver can correctly compensate for delay jitter. The RTP header contains

information that assist the receiver to reconstruct the media and also contains information


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specifying how the codec bit streams are broken up into packets. RTP does not reserveresources in the network but instead it provides information so that the receiver can

recover in the presence of loss and jitter.

The functions provided by RTP include:

Sequencing: The sequence number in the RTP packet is used for detectinglost packets.

Payload Identification: In the Internet, it is often required to change theencoding of the media dynamically to adjust to changing bandwidth

availability. To provide this functionality, a payload identifier is included

in each RTP packet to describe the encoding of the media.

Frame Indication: Video and audio are sent in logical units called frames.To indicate the beginning and end of the frame, a frame marker bit has

been provided.

Source Identification: In a multicast session, we have many participants. Soan identifier is required to determine the originator of the frame. For thisSynchronization Source (SSRC) identifier has been provided.

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

RTCP is a control protocol and works in conjunction with RTP. In a RTP session,

participants periodically send RTCP packets to obtain useful information about QoS etc.

The additional services that RTCP provides to the participants are:

QoS feedback: RTCP is used to report the quality of service. Theinformation 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 toindicate that they are leaving a session.

Identification: Information such as email address, name and phone numberare included in the RTCP packets so that all the users can know theidentities of the other users for that session.

Intermedia Synchronization: Even though video and audio are normally sentover 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.

3.5.Resource Reservation Protocol (RSVP)The network delay and Quality of Service are the most hindering factors in the

voice-data convergence. The most promising, solution to this problem has been

developed by IETF viz., RSVP. RSVP can prioritize and guarantee latency to specific IPtraffic streams. RSVP enables a packet-switched network to emulate a more deterministiccircuit switched voice network. With the advent of RSVP, VOIP has become a reality

today. With RSVP enabled, we can accomplish voice communication with tolerable

delay on a data network. RSVP requests will generally result in resources being reservedin each node along the data path. RSVP requests resources in only one direction;

therefore it treats a sender as logically distinct from a receiver, although the same

application process may act as both a sender and a receiver at the same time. RSVP is not


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itself a routing protocol; it is designed to operate with current and future unicast andmulticast routing protocols. In order to efficiently accommodate large groups, dynamic

group membership, and heterogeneous receiver requirements, RSVP makes receivers

responsible for requesting a specific QoS. A QoS request from a receiver host applicationis passed to the local RSVP process. The RSVP protocol then carries the request to all the

nodes along the reverse data path to the data source. RSVP has the following attributes. It is receiver oriented It supports both unicast and multicast It maintains soft state in routers and hosts, providing graceful support for

dynamic membership changes

It provides transparent operation through routers that do not support it3.6.H.323 Signaling Protocol

H.323 is a standard that specifies the components, protocols and procedures thatprovide multimedia communication services such as real-time audio, video, and data

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

H.323 is part of a family of ITU-T recommendations called H.32x that providesmultimedia communication services over a variety of networks that provide a non-

guaranteed Quality of Service (QOS). This recommendation is based on the real-time

protocol/real-time control protocol (RTP/RTCP) for managing audio and video signals.

One of the primary goals in the development of the H.323 standard was theinteroperability 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.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. These components are:

TerminalsGatewaysGatekeepersMultipoint Control Units (MCUs)


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3.6.1. H.323 TerminalsUsed for real-time communications, an H.323 terminal can be either a personal

computer or a stand-alone device, running an H.323 and the multi-media applications. Itsupports audio communications and optionally supports video and data communications.

The primary goal of H.323 is to inter work with other multimedia terminals. H.323

terminals are compatible with H.324 terminals on switched circuit networks and wirelessnetworks, H.320 on ISDN and H.322 terminals on guaranteed QoS LANs. H.323terminals may be used in multipoint conferences.

3.6.2.H.323 GatewaysA gateway connects two dissimilar networks. A H.323 gateway provides

connectivity between an H.323 network and a non-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 beachieved 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, andH.225 registration, admissions and status (RAS), for registration with the gatekeeper. Onthe SCN side the gateway runs SCN specific protocols such as ISDN and SS7 protocols.


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The gatekeeper provides the functions of address translation, admissions controland bandwidth control for terminals, gateways and MCUs located within its zone of

control.

Optional gatekeeper functions include call-control signaling, call authorization andcall management.

H.225 is part of the H.323 recommendation and it involves call control messagesincluding signaling, registration and admission, and for the packetization andsynchronization of media streams. The H.225 RAS is used between H.323 endpoints for

the following reasons:

Gatekeeper discovery Endpoint registration End point location Admission control Access tokens

The disadvantage of RAS messaging is that these messages are carried on a RASchannel that is unreliable. Hence RAS message exchange may be associated with

timeouts and retry counts.The gatekeeper discovery process is used by the H.323 endpoints to determine the

gatekeeper with whom the endpoint must register. The process of gatekeeper discoverymay be done statically or dynamically. Endpoint registration is a process used by the

endpoints to join a zone and inform the gatekeeper of the zones transport and alias

address. All endpoints automatically register with a gatekeeper as part of theirconfiguration. Endpoint location is a process by which the transport address of an

endpoint is determined and given its alias name or an E.164 address.

3.6.4.Multipoint Control Units (MCUs)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 MCUmanages conference resources, negotiates between terminals for the purpose ofdetermining the audio or video coder/decoder (CODEC) to use, and may handle the

media stream.

The H.323 protocol is specified so that it interoperates with other networks. Themost popular H.323 interworking is IP telephony, when the underlying network of H.323

is an IP network and the interoperating network is SCN. SCN includes PSTN and ISDNnetworks.


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4. Classes of connectionsUsing internet telephony the user can be connected to other user in different

manner according to the networks of both calling and called users. At a call setup as

general, a calling user has to specify a called user by providing the address of called user.

In IP Telephony service, because the interoperation between SCN and the Internet mightbe required, it should be considered how a calling user could specify a called user. Thedifferent type of connection can be.

Telephone terminal

A usual telephone terminal is an endpoint in SCN. A telephone terminal can dealwith the audio and control signals in SCN. E.164 number is assigned to a telephone

terminal in SCN. A calling user can input the digits of 0-9 and the symbols of star (*)

and square (#) through the telephone terminal.Server of IP Telephony

The server of IP Telephony is connected to the Internet and provides the necessary

functions for IP Telephony service, such as the authentication of user, the billing to user,

the identification of the destination address, the selection of the paths to the destination,and so on.

The network of calling user and The network of called user

The local network in which a calling user resides is The network of calling user.Similarly, the local network in which a called user resides is The network of called

user.

The gateway of calling user and The gateway of called userThe gateway of calling user is defined as the gateway connected with the

network of calling user. Similarly, the gateway of called user is defined as the gateway

connected with the network of called user.

4.1.Phone-to-Phone via the InternetIn this connection a calling user on telephone terminal in SCN is connected to acalled user on telephone terminal in SCN via the Internet. Both the network of calling

user and the network of called user are in SCN and have the interconnection with the

Internet. The gateway of calling user and that of called user are required for thisconnection.

A calling user is assigned E.164 number in SCN. The gateway of calling user is

assigned E.164 number in SCN and IP address in the Internet. The gateway of calleduser is assigned IP address in the Internet and E.164 number in SCN. A called user is

assigned E.164 number in SCN.


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The path in this connection is considered as composed of the following three paths.The first path is set from a calling user to the gateway of calling user in SCN, in

accordance with E.164 number. The second path is set from the gateway of calling user

to the gateway of called user via the server of IP Telephony in the Internet, in accordancewith IP address. The third path is set from the gateway of called user to a called user in

SCN, in accordance with E.164 number.

In Phone-to-Phone (Class 1) connection, because a calling user and a called user

are in SCN, a calling user can specify a called user by dialing the E.164 number assignedto a called user. E.164 number is actually used to specify a called user in the present

available service of Phone-to-Phone (Class 1) connection.

This type of connection is most commonly used as a replacement for long distanceor international call service in the traditional telephone connection.

4.2.Pone-to-PC in the InternetIn this connection a calling user on telephone terminal in SCN is connected to a

called user on computer terminal in the Internet. The network of calling user is in SCN,

while the network of called user is the Internet. The gateway of calling and called users

is required for this connection at the interconnection point between the network of calling

user and the Internet.A calling user is assigned E.164 number in SCN. The gateway of calling and

called users is assigned E.164 number in SCN and IP address in the Internet. A called

user is assigned IP address in the Internet. The computer terminal of called user may bedirectly connected to the Internet or connected to the Internet by the dial-in access

through SCN. The called user needs to have their computer terminal powered up andready to receive a call.

The path in this connection is considered as composed of the following two paths.

The first path is set from a calling user to the gateway of calling and called users in SCN,in accordance with E.164 number. The second path is set from the gateway of calling

and called users to a called user via the server of IP Telephony in the Internet, in

accordance with IP address.

Network of

calling userNetwork of

called user

Internet

SCNGateway

of

called user Called

user

SCNGateway

of

calling

user

Calling

user

Server of

IP Telephony

IP Telephony Class1 connection

(Phone-to-Phone via the Internet)


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In Phone-to-PC (Class 2-1) connection, a calling user is in SCN and a called user is

in the Internet. The address information of called user in the Internet, such as IP address,

domain name, e-mail address and so on, does not work out in the addressing system of

SCN. The practical solution to Phone-to-PC (Class 2-1) connection is not yet available.

4.3.PC in the Internet-to-PhoneIn this connection a calling user on the computer terminal in the Internet is

connected to a called user on telephone terminal in SCN. The network of calling user is

the Internet, while the network of called user is in SCN. The gateway of calling and

called users is required for this connection at the interconnection point between the

Internet and the network of called user.A calling user is assigned IP address in the Internet. Like the computer terminal of

called user in Phone-to-PC (Class 2-1) connection, the computer terminal of calling userin this connection may be directly connected to the Internet or connected to the Internet

by the dial-in access through SCN using a modem. The gateway of calling and called

users is assigned IP address in the Internet and E.164 number in SCN. A called user isassigned E.164 number in SCN.

The path in this connection is considered as composed of the following two paths.

The first path is set from a calling user to the gateway of calling and called users via theserver of IP Telephony in the Internet, in accordance with IP address. The second path is

set from the gateway of calling and called users to a called user in SCN, in accordance

with E.164 number.In PC-to-Phone (Class 2-2) connection, a calling user is in the Internet and a calleduser is in SCN. In the present available service of PC-to-Phone (Class 2-2) connection,

the ITSP prepares the database for mapping between E.164 numbers and IP addresses so

that a calling user in the Internet could specify a called user in SCN by inputting theE.164 number.

Network of

calling user

Network ofcalled user

SCN

Internet

Gateway of

calling and called users

Calling

user

Calleduser

Server of

IP Telephony

IP Telephony Class2-1 connection

(Phone-to-PC in the Internet)


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4.4.PC-to-PC in the InternetIn this connection a calling user and a called user are both on the computer terminal

in the Internet. A calling user and a called user are assigned IP addresses in the Internet

respectively. The computer terminal of calling user and that of the called user in this

connection may be directly connected to the Internet or connected to the Internet by thedial-in access through SCN using a modem. The called user needs to have their computer

terminal powered up and ready to receive a call.

Because the network of calling and called users is the Internet, no gateway isrequired for this connection. The path from the calling user to the called user is set

within the Internet, in accordance with IP address. On the other hand, the server of IP

Telephony is required even in this connection for the necessary arrangement for IPTelephony, such as the authentication of user, the billing to user, the identification of the

destination address, the selection of the paths to the destination and so on.

In PC-to-PC (Class 3) connection, because a calling user and a called user are in

the Internet, a calling user can specify a called user by using IP address or other form ofaddress information, such as domain name, e-mail address and so on. In the present

available service of PC-to-PC (Class 3) connection, a calling user specifies a called user

by directly inputting IP address or using the address information through the directoryservice.

IP Telephony Class2-2 connection

(PC in the Internet-to-Phone)

Gateway of

calling and called user

SCN

Internet

Calling

user

Called

user

Server of

IP Telephony

Network of calling user

Network of

called user

InternetCalled

userCalling

user

Server of IP Telephony

IP Telephony Class3 connection

(PC-to-PC in the Internet)

Network of calling user and called user


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4.5.PC in the Internet-to-PC in the separate IP-based network viaSCN

In this connection the path is set between the users on computer terminals via SCN.

Although both a calling user and a called user are on computer terminals, the network of

calling user is in the Internet while the network of called user is the other independent IP-based network separate from the Internet. The network of called user does not have the

direct connection to the Internet and can be connected with the Internet only via SCN.

The addressing/routing system of the network of called user is IP-based but proprietary. Itis independent from the IP address of the Internet. This connection requires the gateway

of calling user at the interconnection point between the Internet and SCN and the gateway

of called user at the interconnection point between SCN and the network of called user.A calling user is assigned IP address in the Internet. Like the computer terminal of

the calling user in PC-to-Phone (Class 2-2) connection, the computer terminal of thecalling user in this connection may be directly connected to the Internet or connected to

the Internet by the dial-in access through SCN using a modem. The gateway of callinguser is assigned IP address in the Internet and E.164 number in SCN. The gateway of

called user is assigned E.164 number in SCN and IP-based but proprietary address in the

network of called user. A called user is also assigned IP-based but proprietary address inthe network of called user.

The path in this connection is consider as composed of the following three paths.

The first path is set from a calling user to the gateway of calling user via the server of IPTelephony in the Internet, in accordance with IP address. The second path is set from the

gateway of calling user to the gateway of called user in SCN, in accordance with E.164

number. The third path is set from the gateway of called user to a called user in the IP-based network of called user, in accordance with the proprietary addressing system of the

network of called user.

In comparison with the path from a calling user to a called user set in PC-to-Phone

(Class 2-2) connection, the path from a calling user to the gateway of called user in thisconnection has the following similar characteristics.

Either path is set through the Internet and SCN.

Internet

Called

user

Calling

user

Server of

IP Telephony

IP Telephony Class 4-1 connection(PC in the Internet-to-PC in the separate IP-based network via SCN)

Gateway of

calling user

Gateway of

called user

SCN

IP-based network


Network of

called user


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Internet

Called

user

Calling

user

Server ofIP Telephony

A part of path in Class 4-1 connection is identical to

the path in PC-to-Phone (Class 2-2) connection

Gateway of

calling user

Gateway of

called user

SCN

IP-based network


Network of

called user

Identical to

PC-to-Phone (Class 2-2) connection(Routing is based on E.164 number.)

(Routing is based on

the proprietary address.)

In either path, a calling user is on computer terminal connected to theInternet. The network of calling user is in the Internet.

The gateway of called user is connected to SCN in this connection while thecalled user is connected to SCN in PC-to-Phone (Class 2-2) connection.

The E.164 number is assigned to the gateway of called user in SCN in thisconnection while the E.164 number is assigned to the called user in SCN inPC-to-Phone (Class 2-2) connection.

Therefore setting the path from a calling user to the gateway of called user in this

connection is considered identical to setting the path from a calling user to a called user

in PC-to-Phone (Class 2-2) connection. The process of setting the path in PC-to-Phone

(Class 2-2) connection can be applied to setting the path from a calling user to thegateway of called user in this connection. After the path from a calling user to the

gateway of called user is set, the path from the gateway of called user to a called user is

set within the IP-based network of called user by the proprietary routing system of that

network.

4.6.PC in the separate IP-based network-to-PC in the Internet viaSCN

In this connection the path is set between the users on computer terminals via SCN.

The network of called user is the Internet while the network of calling user is the separateIP-based network other than the Internet. Like the network of called user in Class 4-1connection, the network of calling user does not have the direct connection to the Internet

and can be connected with the Internet only via SCN. The addressing/routing system of

the network of calling user is IP-based but proprietary. It is independent from the IPaddress of the Internet. This connection requires the gateway of calling user at the

interconnection point between the network of calling user and SCN and the gateway of

called user at the interconnection point between SCN and the Internet.


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A calling user is assigned IP-based but proprietary address in the network of callinguser. The gateway of calling user is assigned IP-based but proprietary address in the

network of calling user and E.164 number in SCN. The gateway of called user is

assigned E.164 number in SCN and IP address in the Internet. A called user is assignedIP address in the Internet. Like the computer terminal of the called user in Phone-to-PC

(Class 2-1) connection, the computer terminal of the called user in this connection maybe directly connected to the Internet or connected to the Internet by the dial-in access

through SCN using a modem.

The path in this connection is considered as composed of the following three paths.The first path is set from a calling user to the gateway of calling user in the IP-based

network of calling user, in accordance with the proprietary addressing system of the

network of calling user. The second path is set from the gateway of calling user to thegateway of called user in SCN, in accordance with E.164 number. The third path is set

from the gateway of called user to a called user via the server of IP Telephony in the

Internet, in accordance with IP address.

In comparison with the path from a calling user to a called user set in Phone-to-PC(Class 2-1) connection, the path from the gateway of calling user to a called user in this

connection has the following similar characteristics.

Either path is set through SCN and the Internet. In either path, a called user is on computer terminal connected to the

Internet.

The network of called user is in the Internet. The gateway of calling user is connected to SCN in this connection while

the calling user is connected to SCN in Phone-to-PC (Class 2-1) connection.

Internet

Called

user

Calling

user

Server of

IP Telephony

IP Telephony Class 4-2 connection(PC in the separate IP-based network-to-PC in the Internet via SCN)

Gateway of

calling user

Gateway of

called user

SCN

IP-based network


Network ofcalled user

Identical to Phone-to-PC (Class 2-1) connection(Routing is basedon the proprietary

address.)


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The E.164 number is assigned to the gateway of calling user in SCN in thisconnection while the E.164 number is assigned to the calling user in SCN in

Phone-to-PC (Class 2-1) connection.

Therefore, setting the path from the gateway of calling user to a called user in this

connection is considered identical to setting the path from a calling user to a called user

in Phone-to-PC (Class 2-1) connection. After the first path is set within the network ofcalling user by the proprietary routing system of that network, the process of setting the

path in Phone-to-PC (Class 2-1) connection can be applied to setting the path from thegateway of calling user to a called user in this connection.

4.7.PC in the separate IP-based network-to-PC in the separate IP-based network via SCN

In this connection the path is set between the users on computer terminals in the

independent IP-based networks separate from the Internet via SCN. Like the network of

calling user in Class 4-2 connection and the network of called user in Class 4-1connection, both the network of calling user and the network of called user do not have

the direct connection to each other as well as the Internet. Both networks can beconnected with each other or the Internet only via SCN. The addressing/routing systemsof both networks are IP-based but proprietary respectively. They are independent from

the IP address of the Internet. This connection requires the gateway of calling user at the

interconnection point between the network of calling user and SCN and the gateway ofcalled user at the interconnection point between SCN and the network of called user.

A calling user is assigned IP-based but proprietary address in the network of calling

user. The gateway of calling user is assigned IP-based but proprietary address in the

network of calling user and E.164 number in SCN. The gateway of called user isassigned E.164 number in SCN and IP-based but proprietary address in the network of

called user. A called user is also assigned IP-based but proprietary address in the

network of calling user.The path in this connection is consider as composed of the following three paths.

The first path is set from a calling user to the gateway of calling user in the IP-basednetwork of calling user, in accordance with the proprietary addressing system of the

network of calling user. The second path is set from the gateway of calling user to the

gateway of called user in SCN, in accordance with E.164 number. The third path is setfrom the gateway of called user to a called user in the IP-based network of called user, in

accordance with the proprietary addressing system of the network of called user.

Like the routing in the network of calling user in Class 4-2 connection, the routingof the first path in this connection is independently conducted by the proprietary routing

system of the network of calling user. Besides, like the routing in the network of called

user in Class 4-1 connection, the routing of the third path is independently conducted bythe proprietary routing system of the network of called user. On the other hand, thesecond path can be set in SCN in accordance with the E.164 number assigned to the

gateway of called user. Setting the second path in this connection is considered identical

to setting the path from a calling user to a called user for a usual telephone service inSCN.


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Thus, although IP Telephony is used in this connection, the ITSP is not involved inthis connection at all. Because this connection does not include the interconnection

between the Internet and SCN, this is not a valid internet telephony connection.

5. Requirements for Internet Telephony ManagementTo manage an IPT network, the enterprise must carefully consider the network

management solutions capabilities and take steps to ensure both short- and long-term

success. These steps can be grouped into 10 requirements. Failing to focus on these islikely to result in failure for the IPT deployment. Each of these requirements and the

associated management capabilities are examined in the following sections.

5.1.Pre-Deployment AssessmentBefore deploying Internet telephony a full network assessment of voice

requirements and the capabilities of the data network is required. Voice packets must be

prioritized above traffic that is not sensitive to delay, such as email and Web traffic.A

data network should support one or more of the QoS protocols designed for delivery ofreal-time data such as voice. At the IP layer, a combination of the QoS protocols like

Resource Reservation Setup Protocol (RSVP) is typically used for voice traffic

prioritization. At the underlying Data Link layer, 802.1p and 802.1q are used to ensureclassification and prioritization of the underlying Ethernet frames containing the voice

packets.

Pre-deployment assessment must include proper capacity planning. Capacityplanning should examine voice requirements by detailing overall source and destination

calling patterns. A historical planning record, if available, will contribute to the best

possible chance of success. The patterns and records should include both station records

and trunk polls. Sophisticated assessment tools will also provide the option to include

Called

user

Calling

user

IP Telephony Class 4-3 connection (PC-to-PC via SCN)

Gateway of

calling user

Gateway of

called user

SCN

IP-based network


Network of

called user

(Routing is based

on the proprietaryaddress.)

IP-based network

(Routing is based

on the proprietaryaddress.)

(Routing is based

on E.164 address.)


20/21


network simulations so that network planners can look at what if scenarios. A completeassessment of voice callers requirements must be matched to the existing or enhanced

data infrastructure.

5.2.Post-Deployment QoS MonitoringOnce deployed, monitoring for all aspects that affect call quality is essential. For

maintaining the QoS some system is required which can keep track of the internettelephony network. Internet telephony management systems should be capable of

monitoring every call as it happens in real-time and providing alerts if voice call routes

and devices are not meeting pre-established QoS metrics.The system should provide post-call QoS recordkeeping, reporting, and analysis

tools. Although the network can be reconfigured to maintain QoS thresholds, the network

operator will want a view of and real-time report of the activity. Periodic macro-levelanalysis is recommended, as part of ongoing operations, and capabilities to drill-down to

a specific callers activity is ideal.

5.3.Performance from the End-User PerspectiveEnd users are accustomed to PSTN quality and will not settle for lesseven for thebenefits of added functionality. Call quality is the key challenge in managing IPTinfrastructure. New features or enhanced functionality will have no value if call quality is

low. Deployment plans, operations, and support tools should be included to provide

proactive support from the users perspective. For example, users have come to expect an

immediate dial tone when they lift the receiver. They expect indications that their callsare being connected within a second or two.

5.4.Managing SecuritySecurity management is an entry-level requirement for any IPT solution. IP-based

phone systems can be as secure as the legacy PSTN; in some cases, the IPT system canbe even more secure. For example, an IPT system can require users to be authenticatedby using a password or identification system in order to make calls and have associated

calling privileges granted to them. Calls can also be encrypted to prevent eavesdropping

attacks.

5.5.TroubleshootingEven with the best pre-planning and highest-available level of fault tolerance, IPT

QoS will experience degradation; therefore, IPT management systems must providetroubleshooting tools. First on the troubleshooting tool list should be an adjustable series

of notification options. Management systems should also provide right-sized alerts.

Too many alerts degrade computing resources and test human patience. Too few alertscan mask small problems that can become high-level failures.

5.6.Automated ManagementWhen building the business case for IPT, every organization will be faced with the

risks and rewards of its proposed network designincluding the cost to the business of

providing device and network redundancy. Some redundancy can be provided inside a


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single device if the device is designed to be fault tolerant; some devices might require anentire backup system.

5.7.Call ManagementCall management, call reporting, and call detail records are vital for any IPT

system. However, the way that the network uses these metrics and how the networksadministrator can use them is what sets an outstanding system apart from an ordinarycall-reporting module. The most advanced systems will integrate with IVR systems and

call directors so that corporate-wide calls can be routed based on callers requests and

employee availability. Some call management systems adopt the call center model thatmatches employee skills to caller profiles and caller service requestsnot only providing

intelligent call management but also supplying another component in workforce

management.

6. ConclusionInternet Telephony is a powerful and economical communication options which is

gaining its popularity, but the most significant obstacles in reaching the height of successis the unsatisfactory voice quality and the lack of means of commercial deployments.

Both of them are under investigation. The voice quality will increase with special QoSmeans and generic increasing bandwidth. Commercial deployment should be designed by

both, commercial and academic world. The standard for addressing a millions of PSTN

user should be made so that they can be able to use it. Simultaneously Internet telephony

systems that are currently deployed should be maintains and managed so that they willencourage others for deploying internet telephony.

7. References.

Reference guide to Internet by M.L.Young. H.323 Version 2 Primer by DataBeam Corporation. IP Telephony Signalling by Bjarne Munch, Ericsson Australia IP Telephony Inter-Gateway Protocols by Alan Percy, Senior Sales Engineer IP Telephony:The Vision, the Reality and the Captaris Role in this Emerging

Market by Captaris.

IP Telephony Management: The Essential Top-10 Checklist by integratedresearch group

International Telecommunications Union ENUM Page,http://www.itu.int/osg/spu/enum/ index.html

ENUM Forum, http://www.enum-forum.org/links.html http://www.databeam.com http://www.intel.com http://www.ietf.org http://www.itu.ch

internet telephony - report

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