Beruflich Dokumente
Kultur Dokumente
On
“Internet Telephony”
Presented by
Nitin Prakash Sharma
M.Tech. IT 1st year
Indian Institute of Technology, Kharagpur
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-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.
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 circuit-
switched 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.
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-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 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.
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.
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
•
Internet Telephony 6/19
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.
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.
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.
Server of
IP Telephony
Internet
Gateway Gateway
SCN of SCN
of
Calling Network of calling called user Network of Called
user calling user user called user user
Network of
calling user Gateway of
SCN calling and called users
Calling
user Network of
called user
Internet
Server of
IP Telephony
Called
user
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.
Server of IP Telephony
Internet Called
Calling
user user
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.
Internet Telephony 13/19
Server of
IP Telephony Internet
Calling
user IP-based network Called
Gateway of user
Gateway of called user Network of
calling user called user
SCN
Server of
IP Telephony Internet
Calling
user IP-based network Called
Gateway of user
Gateway of Network of
called user
calling user called user
SCN
network.
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 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.
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.
(Routing is based
(Routing is based (Routing is based
on the proprietary
on the proprietary on E.164 address.)
address.)
address.)
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.
5.5. 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.
single device if the device is designed to be fault tolerant; some devices might require an
entire backup system.
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