A Seminar

On
“Internet Telephony”

Presented by
Nitin Prakash Sharma
M.Tech. IT 1st year
Indian Institute of Technology, Kharagpur

Under the guidance of

Dr. S.K. Ghosh
Assistant Professor
School of Information Technology
Indian Institute of Technology, Kharagpur
Contents
Abstract.............................................................................................................................. 1
1. Introduction............................................................................................................... 1
2. What is internet telephony (IP Telephony)? .......................................................... 1
2.1. 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 Control
Protocol)........................................................................................................................ 3
3.5. Resource Reservation Protocol (RSVP).......................................................... 4
3.6. H.323 Signaling Protocol.................................................................................. 5
3.6.1. H.323 Terminals........................................................................................ 6
3.6.2. 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-based
network 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
Internet Telephony 1/19
Abstract
Communication 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 hasn’t. 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 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). 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. Introduction
IP 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 launched
research 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 PCs
are 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-touse
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
Internet Telephony 2/19
embodies much more than cheaper long distance calls for friends and families. By
textbook 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).
Here it does not matters whether traditional telephony devices, multimedia PCs or
dedicated 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 a
dependable 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-switched
communication 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 telephony
The 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 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.
3.1.RAS signalling
The RAS channel is a User Datagram Protocol (UDP)- based protocol that is used
for Endpoint Registration/ Deregistration, Admission Control, Bandwidth Change
Request, and Endpoint Status Control. An endpoint can broadcast a request for a
Gatekeeper 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 be
submitted, stating the called endpoint address and requested bandwidth.
Internet Telephony 3/19
The RAS channel provides the means to control user access to the network and
usage of the network.
3.2.Q.931 signalling
The 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 of
connection-oriented communication. It provides capabilities for handling a variety of
supplementary services related to specific connections or users and enables interworking
with the SCN.
3.3.H.245 signalling
The 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 the
Gatekeeper 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 flexible
bidirectional negotiation. These messages provide the means to negotiate different media
formats in each direction, and they can include several media channels in each direction
per call. H.245 is also used to carry Dual Tone Multi Frequency (DTMF) tones end-toend.
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 to
encompass 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 assumes
that 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, Call
Waiting, etc. These services are handled via the call signalling channel, which conveys
the 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 on
the 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 Gatekeeper
control 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-time
Control 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 allow
the 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
Internet Telephony 4/19
specifying how the codec bit streams are broken up into packets. RTP does not reserve
resources 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 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
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. 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.
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. 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.
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 IP
traffic streams. RSVP enables a packet-switched network to emulate a more deterministic
circuit 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 reserved
in 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
Internet Telephony 5/19
itself a routing protocol; it is designed to operate with current and future unicast and
multicast 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 application
is 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 it
3.6.H.323 Signaling Protocol
H.323 is a standard that specifies the components, protocols and procedures that
provide multimedia communication services such as real-time audio, video, and data
communications over packet networks, including Internet Protocol (IP) based networks.
H.323 is part of a family of ITU-T recommendations called H.32x that provides
multimedia communication services over a variety of networks that provide a nonguaranteed
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 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.
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:
• Terminals
• Gateways
• Gatekeepers
• Multipoint Control Units (MCU’s)
•
Internet Telephony 6/19
3.6.1. H.323 Terminals
Used 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. It
supports 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 wireless
networks, H.320 on ISDN and H.322 terminals on guaranteed QoS LAN’s. H.323
terminals may be used in multipoint conferences.
3.6.2. H.323 Gateways
A 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 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.
Internet Telephony 7/19
3.6.3. 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. Gatekeepers in H.323 networks are optional but if they are present
in a network then their services have to be used by the gateways. The H.323 standards
both define mandatory services and optional services that the gatekeeper supports. The
mandatory services of the gatekeeper include address translation, admission control,
bandwidth control, and zone management.
The address translation function of the gatekeeper translates E.164 telephone
numbers (e.g. 214-768-1234) or the alias into the network address (e.g. 47.41.56.123 for
IP) for the destination terminal. Calls that originate within a H.323 network may use an
alias to address the destination terminal, whereas calls that originate outside the H.323
network and received by a gateway may use the E.164 telephone number to address the
destination terminal. The destination end point can be reached using the network address
on the H.323 network
The gatekeeper also performs admission control of the end points into the H.323
network. It uses RAS messages, admission request (ARQ), confirm (ACF), and reject to
achieve this. The admission control could also be a null function that admits all end
points on to the H.323 network
The gatekeeper also provides support for bandwidth control by using the RAS
messages, bandwidth request (BRQ), confirm (BCF), and reject (BRJ). The way this
works is that if a network manager has specified a threshold for the number of
simultaneous connections on the H.323 network, the gatekeeper can refuse additional
connections once that threshold limit has been reached.
Internet Telephony 8/19
The gatekeeper provides the functions of address translation, admissions control
and bandwidth control for terminals, gateways and MCUs located within its zone of
control.
Optional gatekeeper functions include call-control signaling, call authorization and
call management.
H.225 is part of the H.323 recommendation and it involves call control messages
including signaling, registration and admission, and for the packetization and
synchronization 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 RAS
channel 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 discovery
may 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 their
configuration. 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 MCU
manages conference resources, negotiates between terminals for the purpose of
determining 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. The
most 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 ISDN
networks.
Internet Telephony 9/19
4. Classes of connections
Using 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 might
be required, it should be considered how a calling user could specify a called user. The
different type of connection can be.
Telephone terminal
A usual telephone terminal is an endpoint in SCN. A telephone terminal can deal
with 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 user”
“The 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 Internet
In this connection a calling user on telephone terminal in SCN is connected to a
called 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 this
connection.
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 called
user is assigned IP address in the Internet and E.164 number in SCN. A called user is
assigned E.164 number in SCN.
Internet Telephony 10/19
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 accordance
with 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 assigned
to 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 distance
or international call service in the traditional telephone connection.
4.2.Pone-to-PC in the Internet
In 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 be
directly 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 and
ready 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 user
Network of
called user
Internet
SCN
Gateway
of
called user Called
user
SCN
Gateway
of
calling
user
Calling
user
Server of
IP Telephony
IP Telephony Class1 connection
(Phone-to-Phone via the Internet)
Internet Telephony 11/19
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-Phone
In 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 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 calling and called
users is assigned IP address in the Internet and E.164 number in SCN. A called user is
assigned 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 the
server 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 called
user 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 the
E.164 number.
Network of
calling user
Network of
called user
SCN
Internet
Gateway of
calling and called users
Calling
user
Called
user
Server of
IP Telephony
IP Telephony Class2-1 connection
(Phone-to-PC in the Internet)
Internet Telephony 12/19
4.4.PC-to-PC in the Internet
In 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 the
dial-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 is
required 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 IP
Telephony, 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 of
address 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 directory
service.
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
Internet Called
user
Calling
user
Server of IP Telephony
IP Telephony Class3 connection
(PC-to-PC in the Internet)
Network of calling user and called user
Internet Telephony 13/19
4.5.PC in the Internet-to-PC in the separate IP-based network via
SCN
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 IPbased
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. It
is 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 the
calling 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 calling
user 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 in
the 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 IP
Telephony 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 IPbased
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 this
connection 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 calling user
Network of
called user
Internet Telephony 14/19
Internet
Called
user
Calling
user
Server of
IP 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 calling user
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 the
Internet. The network of calling user is in the Internet.
• The gateway of called user is connected to SCN in this connection while the
called 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 this
connection while the E.164 number is assigned to the called user in SCN in
PC-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 the
gateway 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 via
SCN
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 separate
IP-based network other than the Internet. Like the network of called user in Class 4-1
connection, 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 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 of
called user at the interconnection point between SCN and the Internet.
Internet Telephony 15/19
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 is
assigned E.164 number in SCN and IP address in the Internet. A called user is assigned
IP 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 may
be 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 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 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 of calling user
Network of
called user
(Routing is based Identical to Phone-to-PC (Class 2-1) connection
on the proprietary
address.)
Internet Telephony 16/19
• The E.164 number is assigned to the gateway of calling user in SCN in this
connection 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 of
calling 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 the
gateway of calling user to a called user in this connection.
4.7.PC in the separate IP-based network-to-PC in the separate IPbased
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 4-1
connection, 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 be
connected with each other or the Internet only via SCN. The addressing/routing systems
of 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 of
called 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 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 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-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 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.
Like the routing in the network of calling user in Class 4-2 connection, the routing
of 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 by
the proprietary routing system of the network of called user. On the other hand, the
second 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 in
SCN.
Internet Telephony 17/19
Thus, although IP Telephony is used in this connection, the ITSP is not involved in
this 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 Management
To manage an IPT network, the enterprise must carefully consider the network
management solution’s 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 is
likely 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 Assessment
Before 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 of
real-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 ensure
classification and prioritization of the underlying Ethernet frames containing the voice
packets.
Pre-deployment assessment must include proper capacity planning. Capacity
planning 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 calling user
Network of
called user
(Routing is based
on the proprietary
address.)
IP-based network
(Routing is based
on the proprietary
address.)
(Routing is based
on E.164 address.)
Internet Telephony 18/19
network simulations so that network planners can look at “what if” scenarios. A complete
assessment of voice callers’ requirements must be matched to the existing or enhanced
data infrastructure.
5.2.Post-Deployment QoS Monitoring
Once 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 internet
telephony 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-level
analysis is recommended, as part of ongoing operations, and capabilities to drill-down to
a specific caller’s activity is ideal.
5.3.Performance from the End-User Perspective
End users are accustomed to PSTN quality and will not settle for less—even for the
benefits of added functionality. Call quality is the key challenge in managing IPT
infrastructure. 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 calls
are being connected within a second or two.
5.4.Managing Security
Security 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 can
be even more secure. For example, an IPT system can require users to be authenticated
by 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.Troubleshooting
Even with the best pre-planning and highest-available level of fault tolerance, IPT
QoS will experience degradation; therefore, IPT management systems must provide
troubleshooting 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 alerts
can mask small problems that can become high-level failures.
5.6.Automated Management
When building the business case for IPT, every organization will be faced with the
risks and rewards of its proposed network design—including the cost to the business of
providing device and network redundancy. Some redundancy can be provided inside a
Internet Telephony 19/19
single device if the device is designed to be fault tolerant; some devices might require an
entire backup system.
5.7.Call Management
Call 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 network’s
administrator can use them is what sets an outstanding system apart from an ordinary
call-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 that
matches employee skills to caller profiles and caller service requests—not only providing
intelligent call management but also supplying another component in workforce
management.
6. Conclusion
Internet Telephony is a powerful and economical communication options which is
gaining its popularity, but the most significant obstacles in reaching the height of success
is 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 QoS
means 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 will
encourage 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 integrated
research 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