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Presented by
Nitin Prakash Sharma
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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.
