IP TELEPHONY: AN INTRODUCTION

BY ENGR. KHALIL AHMED

Royal Saudi Naval Forces
Riyadh, Saudi Arabia

ABSTRACT
IP telephony is a rapidly emerging technology for voice communication that has astonished both the data-
communication and telecom industries. Technical developments over last few years have made the use of IP networks for
telephony applications a reality. The objective of this paper is to provide the introduction to the technology, and describe
the related protocols and the issues behind delivering an “appropriate” quality of service.
The paper reviews the IP telephony technology, covering its background, working, and various methods of its
implementation. It then explores the pros and cons of this exciting technology and looks at what the future has to offer. It
also examines the market trends and looks at what market researchers think about IP telephony. The paper also covers the
architecture and protocols underlying IP telephony, and discusses the various signaling standards that make the
technology possible. At the end, it covers the issues related to the voice quality and discusses how these may be dealt with.

INTRODUCTION
ittle more than a decade ago Internet was not caught the world's attention. The technology has now

L available to general public, interactive voice

communications were only made by telephone at
Public Switched Telephone Network (PSTN) and the data
exchange was expensive for long distances. Then we saw
improved to a point where conversations are easily possible
and it continues to get better. Technology is improving so
fast that it will not be long until phone-to-phone service on
the Internet offers the same call quality as conventional
appearing some interesting things: Personal Computers to phone service.
large masses and Internet to general public. People begun
to use PCs and Internet to communicate and to exchange IP TELEPHONY
data (images, sounds, documents) with new services like
email, chat and discussion groups. IP telephony (also known as Internet telephony or Voice
Availability of a telephone and access to a high quality over IP) is transport of voice calls over packet-switched IP-
public switched telephone network is considered to be based data networks, no matter whether traditional
essential in modern society. There is however, a Paradigm telephony devices, multimedia PCs or dedicated terminals
shift beginning to occur since more and more take part in the calls and no matter whether the calls are
communications is in digital form and transported via entirely or only partially transmitted over the IP network.
packet networks such as IP, ATM and Frame Relay. Since This technology enables standard data packets to transmit
data traffic is growing much faster than telephone traffic, multimedia information such as voice, fax or video over the
there has been considerable interest in transporting voice Internet or a corporate intranet (IP-based private network)
over data networks. with suitable quality of service (QoS) and a much superior
Support for voice communications using the Internet cost/benefit ratio. It draws on open standards and
Protocol (IP), usually called “IP telephony”, has become recommendations generated by international groups such as
especially attractive given the low cost, flat-rate pricing of the Internet Engineering Task Force (IETF) and the
the Internet. In fact, toll quality telephony over IP has now International Telecommunication Union (ITU). All
become one of the key steps leading to the convergence of suppliers of Internet telephony products meet these
the voice, video and data communications industries. standards.
The first IP telephony product was introduced by When used only across an IP network, such as intranet
Vocaltec in early 1995. In a very short period of time it has or Local Area Network (LAN), it is generally known as

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 IP Telephony: Technology and Standards

“Voice over IP” (VoIP). When the call originates and/or Packet Switched Connection: While circuit switched
terminates in public switched telephone network, and the connection is open and constant for the entire duration of
transport is Internet or Wide Area Network (WAN), it is the call, packet switched connection opens just long enough
generally called “IP telephony” or “Internet telephony.” to send a small chunk of data, called a packet, from one
IP telephony requires people, who want to talk to each system to another. A packet switched connection keeps you
other, to log onto a computer equipped with a microphone connected all the time but you only pay for the amount of
and speaker and establish a connection over the Internet. data transferred. In this case, the data is divided into small
However, a user doesn't have to be online to reap the packets and each packet contains a source and a destination
benefits of IP telephony – A user logged on to a computer address. Packets of data are sent from source to destination
may also make a call to a telephone. Even a telephone to using the quickest route available. The network bandwidth
telephone call can be made over the Internet. Whenever a is shared and multiple simultaneous users are allowed to
telephone is used, the call must be transferred from the access multiple locations across a network. This provides
Internet to the local telephone system. The companies that for much more efficient use of available bandwidth but can
provide Internet phone software also provide gateways create problems for voice traffic, which is very sensitive to
through which these conversions occur. A fee for using the delay.
gateway is incurred by the user; these charges are very low The PSTN is a circuit-switched network that has been
compared to standard long-distance phone call charges. For optimized for real-time or synchronous voice
example, a call to an ordinary telephone in United States communication with a guaranteed Quality of Service
from Saudi Arabia over the Internet could be as low as (QoS). The PSTN guarantees the QoS by dedicating a full-
$.04/minute, as compared to $0.80/minute over telephone duplex 64Kbps circuit between the parties of a telephone
lines. Today, users can bypass long-distance carriers and conversation. Regardless of whether the parties are
their per-minute usage rates and run their voice traffic over speaking or silent, they use full 64K dedicated circuit until
the Internet for a flat monthly Internet-access fee. the call ends. Much of this capacity is wasted during a
Rapidly changing technology is making IP telephony a normal telephone conversation, because while the line is
legitimate alternative for voice services over the Internet. working at full capacity, not all of each user’s time is spent
And it doesn’t stop with just voice over the Internet – other transferring data or talking.
applications include fax over the Internet, call center On the other hand, IP networks are packet-switched
integration, conference bridging and telecommuter access. networks that have historically been used for applications
These applications are all made possible over a single where a variable QoS is tolerable. Instead of keeping a
access line using TCP/IP architecture and the Internet. circuit open constantly, IP networks send and receive data
only as needed, a bit at a time, in data packets. By doing so,
PRINCIPLES OF IP IP networks free up network resources, as well as the
resources of the computers sending and receiving
TELEPHONY information. Since these do not dedicate a path between
sender and receiver, these cannot guarantee QoS. There are,
To understand IP telephony, it’s necessary to be however, ways that can be used to get reasonable QoS on
familiar with the fundamental characteristics behind the the IP networks.
Internet and how it compares to the Public Switched
Telephone Network (PSTN). The most important of these
characteristics is the data transport mode, also known as
The Basics of IP Voice Call
data connection type which is either a circuit switched or IP telephony technology uses packet-switching to
packet switched as explained below: minimize the amount of resources used in a telephone
Circuit Switched Connection: A device using a circuit connection. The telephony application digitizes and
switched connection only connects when data is to be sent. compresses the analog voice signals. This data is then
The connection is dedicated exclusively to the sending and transmitted as a stream of packets over an IP network. IP
receiving nodes for the entire duration of the call. Because network allows each packet to independently find the most
the two points are connected in both the directions, the efficient path to the intended destination, thereby best using
connection is called a circuit. The connection is only the network resources at any given instant. At the
present when you need it and, since bandwidth remains destination, the packets are re-assembled back into their
constant, you only pay for the duration of the connection. original order. The recipient IP telephony application then
While connected on a circuit switched network you have decompresses the packets and converts them back into the
exclusive use of the established connection and data can be analog voice signal. The application insures proper
sent continuously. This type of data transaction is typically reconstruction of the voice signals, compensating for
routed through the PSTN. Although the circuit switched echoes, jitter, and for dropped packets. The actual end-to-
network provides a very reliable connection for voice end process, however, may involve more steps as described
transmissions, it makes very inefficient use of the available below:
bandwidth.

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 IP Telephony: Technology and Standards

IP Phone
Multimedia PC Digital Voice Multimedia PC
Packets
with Phone Software with Fax Software

` LOCATION D `

LOCATION C LOCATION E

IP Telephony IP Network IP Telephony

Gateway (Internet, intranet, etc.) Gateway

PSTN PSTN

IP Phone Server
Analog Voice for authentication, accounting Analog Voice
Signal Signal
and routing
Fax Phone Fax Phone

LOCATION A LOCATION B

Figure 1: IP Telephony Infrastructure

A phone or fax call goes over the local PSTN to the the receiving end decompresses the data and converts it
nearest IP telephony gateway (see Figure 1). The gateway back into analog voice signal using a Digital to Analog
uses an Analog to Digital Converter (ADC) to encode the Converter (DAC). The gateway then hands over the voice
voice digitally and to compress/translate the signal into IP signal to the PSTN network, which in turn passes it on to
packets. It then moves the data onto IP network for the phone or fax at the receiving end.
transport to a gateway at the receiving end. The gateway at

IP TELEPHONY SCENARIOS
he IP telephony usage scenarios, as shown in Figure PC-to-PC calls, except for the cost of Internet access. The

T 1, are commonly classified by the type of devices

terminating an IP call. Because there may be either
a PSTN device (e.g. telephone) or a data-oriented terminal
(e.g. personal computer) on each side of a call, there are
user doesn’t even have to pay for long-distance calls. This
pure-IP scenario can also take advantage of integration with
other Internet services, such as instant messaging, video
conferencing, etc.
four generic classes as below:
PC-TO-TELEPHONE
PC-TO-PC
The PC-to-Telephone method allows a user to call any
PC-to-PC communication can be provided for ordinary telephone on a PSTN from his computer. In this
multimedia PCs (i.e. Personal Computers with a case, a gateway converting the IP call into a PSTN call has
microphone, speaker and a sound card) operating over an to be used. The gateway is required to be located as near to
IP-based network without connecting to the PSTN. PC the called party as possible, to minimize the price for the
applications (and IP-enabled telephones) can communicate gateway-to-called party connection. The call is converted to
using point-to-point or multipoint sessions. This set up a PSTN call at the gateway and is then sent over PSTN to
requires that parties be equipped to talk at the time of the its destination.
call. Like PC-to-PC calling, this scenario requires a software
This class is attractive especially for private users who client. The software is usually free, but the caller may have
already have an Internet access and a multimedia PC. to pay a small per-minute charge to a gateway operator.
Necessary software is available from several companies for The cost charged by the operator is determined mainly by
free or at a very low cost. There is usually no charge for the cost of the call placed from the gateway to the called

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 IP Telephony: Technology and Standards

TELEPHONE-TO-
party. This solution is commercially available from
Net2Phone, PhoneServe and many other companies.

TELEPHONE-TO-PC TELEPHONE
The Telephone-to-Telephone communication appears
The Telephone-to-PC method allows a user to call from
like a normal telephone to the caller but may actually
any ordinary telephone on a PSTN to a PC connected to the
consist of various forms of voice over packet network, all
Internet. In this case, a gateway converting the PSTN call
interconnected to the PSTN. In this scenario, a caller dials
into an IP call has to be used, and the gateway is required to
into a gateway using a regular telephone. The call is
be located as near to the caller as possible. The call is
converted to an IP call at the gateway and the voice data
converted to an IP call at the gateway. The voice data then
“hops on” the Internet. At the end point the voice data hits
“hops on” the Internet and finds the PC on the other end by
another gateway and “hops off” the Internet. The voice data
using the unique IP address.
is converted back to PSTN format and sent over the PSTN
A few companies are providing special numbers or
to its destination.
calling cards that allow a standard telephone user to initiate
This class is attractive for those who want to save on
a call to a computer user. The caveat is that the computer
long-distance call and do not want to use PC. Since the call
user must have the vendor's software installed and running
has to pass through two gateways – PSTN-to-Internet and
on his computer. This solution may require user to pay
Internet-to-PSTN, the cost is charged by both the gateway
local call charges, in addition to small per-minute charge to
operators. In addition, the user may have to pay local call
a gateway operator.
charges. This solution is commercially available from many
companies, offering discounted rates for long distance IP
telephony calls.

BENEFITS AND DRAWBACKS

BENEFITS • Network Efficiency: The sharing of equipment and
operations costs across both data and voice users
IP telephony could be applied to almost any voice can also improve network efficiency since a packet-
communications requirement, ranging from a simple inter- switched IP network can handle more calls with the
office intercom to a complex multi-point teleconferencing same transmission infrastructure than the PSTN can
environment. Listed below are the main benefits of using IP with its circuit switched TDM approach.
telephony: • Integration: Universal use of the IP protocols for
both data and voice applications holds out the
• Cost Reduction: The first measure of success for IP promise of reduced complexity and more flexibility.
telephony is the cost savings for long distance calls. This provides an opportunity to share facilities such
Today flat rate long-distance pricing is available as directory services and security services, and
with the Internet and can result in considerable eliminate points of failure.
savings for both voice and fax. Large organizations • Advanced Applications: In addition to basic
with offices around the world save even more on telephony and fax, the longer term benefits are
long-distance calls by using local Internet gateways. expected to be derived from multimedia and point-
IP telephony provides a competitive threat to of-service applications such as directory services
providers of traditional telephone services that will that enable conference calls to be set up from Web-
clearly stimulate improvements in cost and function based directories, and wireless unified messaging,
throughout the industry. which will let users retrieve their voice and e-mail
• Simplification: IP telephony enables a company to messages via their cellular phones. Combining voice
use a single communications medium rather than and data features into new applications will also
having to maintain separate systems for voice and provide the greatest returns over the longer term.
data communications. An integrated infrastructure
that supports all forms of communication allows DRAWBACKS
more standardization and reduces the total
equipment needs. Listed below are the main issues and drawbacks of
• Flexibility: IP telephony equipment has the using IP telephony:
flexibility to cater to a wide range of configurations
and environments and the ability to blend traditional • Voice Quality: The voice quality of an IP telephone
telephony with IP telephony. Hence, the quality of call over Internet is usually not quite as good as a
voice reproduction can be tailored according to the conventional phone, but the low price compensates
application. Customer calls may need to be of for it. The quality is often compared to that of a
higher quality than internal corporate calls. speakerphone; a little choppy at times, but generally

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 IP Telephony: Technology and Standards

understandable. Calls on the public switched • Standards: The major difficulty that IP Telephony
telephone network usually exhibit 50 to 70 ms technology is facing is the interoperability between
delay. That latency increases substantially on the IP telephony products. Hence, the users who want to
Internet, where it typically ranges to 500 ms. make IP phone call have to have the same kind of
• Capacity: One of the main parameters affecting the software or IP telephony equipment.
quality of service on the Internet is lost packets. It is • Regulation: Traditionally, telephone service has
a persistent problem, particularly with the increasing been heavily regulated. However, regulation of
load of the Internet. This is mainly a function of Internet telephony is still largely a question mark.
network congestion. When traffic causes delays or Internet telephony has stirred fears from carriers
out-of-order packets, some packets are dropped, throughout the globe, many of whom have reacted
causing breaks (silence) in the signal. Inadequate by seeking regulatory protection from the new
network access links, especially local ISP technology. In most countries, governments or
connections to the Internet backbone, are the major government- authorized entities retain the right for
cause for bandwidth congestion. providing telephone service.

APPLICATIONS

I P telephony enables a whole new generation of

applications which are impossible with other telephony
architectures. Some examples of the applications that
are likely to be useful are as follows:
integrated voice/data mailboxes. Voice annotated
documents and multimedia files can easily become
standard within office suites in the near future.
6. Internet Call Center Access: Access customer
service agents online over an Internet call center.
1. Advanced Intelligent Network Features: Use 7. Virtual Call Centers: Support the integrated voice
advanced intelligent network (AIN) features such as and data requirements of call center agents working
Caller ID, voice mail, call waiting, pre-and-post from their homes.
paid calling cards, call blocking, and auto call-back 8. Live Auction Websites: Create live audio auction
in IP telephony. websites for excess inventory. Use Java applets on
2. Voice Calls from Mobile Laptop PCs: Call office the phone to manage the bidding process and to
or home, from hotel, airport, etc. using multimedia track who “raised a hand” to bid first, etc.
laptop PCs with wireless connection to Internet. 9. Presence and Instant Messaging: Use instant
This could be ideal for submitting or retrieving messenger service to determine when
voice messages. geographically distributed colleagues are available
3. Airlines Reservations: Use a Java applet to for a quick conference call with a customer.
visually display interactive voice response options 10. Electronic Business Cards: Send an enriched
rather than forcing users to wait through very long electronic virtual business card (vCard) including
recorded instructions and go through multi-level photo and audio file automatically with every call as
menus requiring the use of a telephone keypad. caller ID information (or selectively during the
4. Internet-aware Telephones: Use enhanced middle of call).
ordinary telephone (wired or wireless) as an Internet 11. Integrated Voice and Data Information: Integrate
access device as well as for normal telephony. voice and data information collected during a call
Directory services, for example, could be accessed with sales force automation applications.
over the Internet by submitting a name and 12. Personalized Music On-hold: Play personalized
receiving a voice (or text) reply. announcements or music from a favorite MP3
5. Voice Annotated Ddocuments: Send voice recording or Internet radio station while callers are
messages and voice annotated documents to on hold.

MARKET TRENDS

I P telephony is becoming a key driver in the evolution

of voice communications. While it is currently a small
fraction of telecommunication market, it is growing
quickly. Here is a selection of what market researchers
think about IP telephony:
substantial growth between 2004 and 2007,
increasing by 48% in 2004 and topping $4.9 billion
by year's end. Equipment sales will continue to
grow by more than 50% in 2005 and 2006, to reach
$11.4 billion in 2006, and $15.1 billion in 2007
(Figure 2).
• According to IDC's January 2003 forecast, • Businesses spent an estimated $2.0 billion on IP-
worldwide sales of IP telephony equipment will see capable telephone systems in North America in

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 IP Telephony: Technology and Standards

• VoIP will account for approximately 75% of world

voice services by 2007. (Frost & Sullivan, March
2002)
• By 2008, wholesale VoIP traffic in the Europe,
Middle East and Asia (EMEA) region could reach
57 billion minutes. About 1,771 billion minutes of
retail voice traffic originating in EMEA will have a
VoIP component in some portion of the route. (Frost
and Sullivan, May 2002)
• According to a study conducted by Insight Research
Figure 2: Worldwide IP Telephony Equipment Corporation, from a mere $13 billion in 2002, voice
Revenues, 2001-2007 (in Billions & % Growth) over packet (VoIP)-based services will grow to
nearly $197 billion by 2007. Also, VoIP services
2003, according to Gartner Dataquest, which also will grow at a compounded rate of over 72 percent
predicts that enterprises spending on IP phone over the period 2002-2007, making packet voice
systems will more than double during four years, to services one of the fastest-growing segments in the
reach $4.2 billion in 2007. telecommunications industry (Press Release –
• According to Gartner, the IP telephone systems “Phone Industry Survival hinges on Explosive
accounted for just 1.4% of total business telephony Growth of VoIP” – October 28, 2002)
equipment sales in 1999. However, in 2003 IP • Enterprises will migrate their voice systems from
telephony equipment increased its market share to traditional networks to data networks at a rate that
56% of all sales, and by 2007, it is expected to will create a $16.5 billion dollar IP-PBX market
account for 97% of all business telephony sales. worldwide by 2006. (Allied Business Intelligence.)
• The IP PBX market is expected to grow to $3.9 • 90% of enterprises with multiple locations will start
billion in revenue by 2005, representing nearly 20% switching to IP systems for voice over next 5 years.
of all traditional PBX sales. (Synergy Research, (Phillips Group, via Aspect, June 2001)
February 2002) • The European IP Virtual Private Network (VPN)
• By 2005, analysts predict 34% of all telephone calls services market to grow from 1.74 billion at the end
made worldwide will be carried over the Internet, of 2002 to 3.58 billion by 2005, when almost 40%
accounting for over 90 billion minutes of telephone of the identified potentials within the European
communication. market will have implemented an IP VPN solution.
• Wholesale and retail VoIP traffic volume exceeded (Frost and Sullivan, January 2003)
6 billion and 15 billion minutes in 2000. VoIP will
account for approximately 75% of world voice
services by 2007. (Frost & Sullivan, May 2001)

IP TELEPHONY PROTOCOL ARCHITECTURE

L ike any conversation

between two
computing devices, IP
telephony requires an agreed
upon set of rules, called
“protocol”. Figure 3 shows an
architectural diagram of the
protocols that are used to
govern IP calls. These
protocols follow a layered
hierarchy which can be
compared with the Open
Systems Interconnect reference
model (OSI 7-layer model),
developed by the International
Standards Organization (ISO).
Although the IP telephony
model doesn’t exactly match
OSI structure, it is useful for
discussion. Figure 3: IP Telephony Protocol Suite in Relation to the ISO Model

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 IP Telephony: Technology and Standards

Breaking a system into defined layers can make that TCP is a connection-oriented protocol. It establishes a
system more manageable and flexible. A layer defines a logical end-to-end connection between the two
specific data communication function that may be communicating hosts before transmitting data. TCP also
performed by any number of protocols. Each layer has its provides a reliable data delivery across the network using a
job, and does not need a detailed understanding of the mechanism called Positive Acknowledgment with Re-
layers around it. For example, IP datagram (or packet) can transmission (PAR). Simply stated, a system using PAR
be transported across a variety of physical layer systems sends the data again, unless it hears from the remote system
including serial lines, Ethernet and ATM. The data is that the data arrived okay.
passed down the stack when it is being sent to the network TCP also handles sequencing and error detection,
and up the stack when it is being received from the ensuring that a reliable stream of data is received by the
network. destination application. It views the data it sends as a
The physical and link layer protocols provide the means continuous stream of bytes, not as independent packets.
for the system to deliver data to the other devices on a Therefore, TCP takes care to maintain the sequence in
directly attached network. Unlike higher-level protocols, which bytes are sent and received.
these protocols must know the details of the underlying Although the TCP works smoothly with data – any
network (its packet structure, addressing, etc.) to correctly packet not delivered is simply re-transmitted, it does not
format the data being transmitted to comply with the work well with real-time voice and video applications. It is
network constraints. The physical layer protocol is for the because any word received out of sequence within the
most part irrelevant to IP and need not be the same for first structure of a sentence will result in a garbled message.
link and final link of a VoIP call.
The major protocols involved in IP telephony, started at USER DATAGRAM
the Network layer are discussed in the following sections:
PROTOCOL (UDP)
INTERNET PROTOCOL (IP)
User Datagram Protocol (UDP) is also a transport layer
The Internet Protocol (IP) is a network layer protocol protocol and is responsible for the transmission of
and it is the heart of the IP telephony. It provides the basic information between the correct applications on the host
packet delivery service on which IP telephony networks are computers. The UDP uses 16-bit source-port and
built. All protocols, in the layers above and below IP, use destination-port numbers in the message header, to deliver
the Internet Protocol to deliver data. data to the correct applications process.
IP is responsible for defining the datagram (or packet) Like IP, UDP is a connectionless protocol as it does not
and moving data between link layer and transport layer. It establish an end-to-end connection between two
is also responsible for routing datagrams to remote hosts communicating hosts before transmitting data. It routes
and performing fragmentation and re-assembly of the data to the correct destination port, but does not attempt to
datagrams. IP is a connectionless protocol, that is, it does perform any sequencing, or to ensure data reliability. The
not establish an end-to-end connection through a network UDP gives application programs direct access to a
before transmitting data. IP relies on protocols in other datagram delivery service, like the delivery service that IP
layers to establish the connection if they require provides. This allows applications to exchange messages
connection-oriented service. over the network with a minimum of protocol overhead.
IP makes no guarantees concerning reliability, flow For voice applications, UDP ensures that the
control, error detection or error correction. This means that information is received in correct sequence, reliably and
the packets could arrive at the destination computer out of with predictable delay characteristics. UDP also performs
sequence, with errors or may not arrive at all. IP just certain functions that TCP cannot perform. Hence, it is
transports the data to a higher layer and relies on protocols commonly used for IP telephony.
in the other layers to provide error detection and error
correction. REAL-TIME TRANSPORT
TRANSMISSION CONTROL PROTOCOL (RTP)
PROTOCOL (TCP) Real time applications, such as voice and video, require
mechanisms to be in place to ensure that a stream of data
The Transmission Control Protocol (TCP) is a transport can be reconstructed accurately. In IP networks, the
layer protocol and is responsible for delivering data connection-oriented TCP protocol guarantees an error-free
received from IP to the correct application. The application transmission in the right order but it is not appropriate for
that the data is bound for is identified by a 16-bit number real-time applications. For instance, with video, if a packet
called the port number (for example, the HTTP application arrives late, it loses its meaning and may not be inserted
is usually associated with port 80). TCP uses 16-bit source- correctly in the clip being played. For this reason, UDP is
port and destination-port numbers in segment header, to used for voice and video transmission.
deliver data to the correct applications process. Jitter is the variation in delay times experienced by the
individual packets making up the data stream. In order to

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 IP Telephony: Technology and Standards

RTP CONTROL PROTOCOL

reduce the effects of jitter, data must be buffered at the
receiving end of the link so that it can be played out at a
(RTCP)
constant rate. However, the UDP has no control over the
order in which packets arrive at the destination or how long
it takes them to get there. Both of these are very important
The RTP Control Protocol is the counterpart of RTP
to overall voice and video quality. Real-time transport
that provides control services. The RTCP monitors the QoS
protocol (RTP) solves the problem by enabling the receiver
and conveys information about the participants in an
to put the packets back into the correct order and not wait
ongoing session. It provides feedback on total performance
too long for packets that have either lost their way or are
and quality so that modifications can be made. Other RTCP
taking too long to arrive. RTP is really just the underlying
functions include carrying a transport-level identifier for an
transport plane, and the other protocols use it to
RTP source which is used by receivers to synchronize
successfully transport voice and video packets.
audio and video.
RTP is designed to provide end-to-end network
RESOURCE RESERVATION
transport functions for applications transmitting real-time
data, such as voice and video, over IP networks. It carries
data source and payload type information, and is itself
carried inside of UDP. RTP provides the sequence number PROTOCOL (RSVP)
and time stamp information needed to assemble a real time The Resource Reservation Protocol is a signaling
data stream from packets. However, RTP does not have any protocol that supports the reservation of resources across an
mechanisms for ensuring the on-time delivery of traffic IP network. It manages quality of service by requesting a
signals or for recovering lost packets. Therefore, RTP does certain amount of bandwidth and latency in every network
not reduce the overall delay of the real time information. hop that supports it. Applications running on IP end
Nor does it make any guarantees concerning quality of systems can use RSVP to indicate to other nodes the nature
service. (bandwidth, jitter, maximum burst, and so forth) of the
packet streams they want to receive. The RSVP depends on
IPv6.

IP TELEPHONY SIGNALING PROTOCOLS

S
everal standards are available for building IP communicate with each other. H.323 is a comprehensive
telephony solutions. These include H.323, Session and very complex protocol. It provides specifications for
Initiation Protocol (SIP), Media Gateway Control real-time, interactive videoconferencing, data sharing and
Protocol (MGCP) and Media Gateway Control (Megaco). IP telephony. The Session Initiation Protocol (SIP)
A high-level comparison of these protocols is included in emerged as an alternative to H.323. It is a much simpler,
Table 1. more streamlined protocol developed specifically for IP
Of the protocols listed in Table 1, only SIP and H.323 telephony. It is smaller and more efficient than H.323 and
are peer-to-peer protocols. MGCP and Megaco represent takes advantage of existing protocols to handle certain parts
the old centralized model and suffer from this model’s of the process.
limitations. Thus, the real choice for a protocol with Web- Both of these protocols do essentially the same things;
like benefits comes down to one of the peer-to-peer these provide a way for the caller to find the called party
protocols – H.323 or SIP. (call construction), allow each party to send streams of
The H.323 and SIP are the two major protocols that are audio data to the other party, and provide a way for either
used by VoIP technology to define ways for devices party to end the call (call tear-down). Both of these
(telephones, computers, etc.) on the data network to protocols also include specifications for audio (and video)

Table1: Comparison of IP Telephony Protocols

H.323 SIP MGCP Megaco/H.248
Standardization ITU-T IETF IETF IETF and ITU-T
Architectural model Peer-to-peer Peer-to-peer Master/ slave Master/ slave
Media types Voice, video, limited data Voice, video, data Voice Voice, video
Call control Gatekeeper Proxy/Redirect Server MGC MGC
Endpoints Gateway, terminal User agent Media gateway Media gateway
Signaling transport TCP or UDP TCP or UDP UDP TCP and UDP
Network scope Intranet, Extranet, Internet Intra-, Extra-, Internet Intranet only Intranet only
Extensibility Low High Medium Medium
Scalability Medium High Low Low
Ease of deployment Low High Medium Medium
Standardization ITU-T IETF IETF IETF and ITU-T

78 IEP-SAC Journal 2004-2005

 IP Telephony: Technology and Standards

codecs (coder-decoder) to convert audio signal into a introducing only a few new powerful features to the base
compressed digital form for transmission and back into an document. The last Version 5 was approved at the end of
uncompressed audio signal for replay. These protocols are July 2003. Unlike previous revisions of the
discussed in detail in the following sections: recommendation, Version 5 aimed to maintain stability in
the protocol by introducing only modest additions to the
H.323 base protocol, rather than introducing sweeping changes as
was the case in prior revisions.
The H.323 is an umbrella recommendation from the
Telecommunication Standardization Sector of the H.323 Architecture
International Telecommunications Union (ITU-T) that
specifies the components, protocols and procedures that
Components
provide multimedia communication services (real-time The H.323 standard specifies a number of components
voice, video, chat, whiteboard, file sharing, etc.) over (entities), which, when networked together, provide the
packet–based networks, including IP. By conforming to point-to-point and point-to-multipoint multimedia-
H.323 standards, multimedia products and applications communication services. Some components are mandatory,
from different vendors can interoperate across IP based while others are optional. The four most important
networks, including the Internet. H.323 is part of a family components are listed below:
of ITU-T recommendations called H.32x that provides
multimedia communication services over a variety of Terminal: An H.323 terminal is an endpoint on a network
networks. which provides two-way communications with another
H.323 covers both protected and unprotected terminal, gateway or a Multipoint Control Unit (MCU). An
connections. Control and data information requires a H.323 terminal can either be a personal computer or a
protected transmission to prevent packets from being lost or stand-alone device such as IP telephone running H.323 and
not received in the right order. In IP-based networks, TCP the multimedia applications. It supports audio
protocol guarantees an error-free transmission in the right communications and can optionally support video or data
order but causes delays and has a lower throughput. communications.
Therefore, unprotected connections are used for audio and
video transmissions, which are more efficient. Gateway: An H.323 gateway provides connectivity
The H.323 standard's mandatory components are between an H.323 network and a non–H.323 network. For
transmission of audio, connection control according to example, it may route Voice over IP (VoIP) calls from an
Q.931, communication with the gatekeeper over the RAS H323 terminal to the public switched telephone network
protocol, and use of the H.245 signaling protocol; the rest (PSTN). This connectivity of dissimilar networks is
of the text, including coverage of the ability to transmit achieved by translating protocols for call setup and release,
video and data, is optional. Although H.323 uses TCP to converting media formats between different networks, and
carry the signaling channels, the real-time media streams transferring information between the networks connected
are transported on RTP/RTCP (discussed earlier). RTP by the gateway. A gateway is not required, however, for
carries the actual media and RTCP carries status and communication between two terminals on an H.323
control information. network.
Being the first widely available VoIP protocol, H.323
enjoyed a head start as developers implemented it as toll- Gatekeeper: A gatekeeper provides basic admission
bypass systems as well as PC-to-phone and video- control onto a network by allowing or refusing
conferencing applications. The best-known H.323 communications between other H.323 entities within its
application was Microsoft NetMeeting. zone of control. They also provide call-control services for
H.323 endpoints, such as address translation (to use name
History instead of IP address), authentication, accounting and
The Version 1 of the H.323 recommendation was bandwidth management. Gatekeepers in H.323 networks
accepted in October 1996. It was heavily weighted towards are optional. If they are present in a network, however,
multimedia communications in LAN environment and does terminals and gateways must use their services.
not provide guaranteed quality of service. With the
development of VoIP, new requirements emerged, such as Multipoint Control Unit (MCU): An MCU provide
providing communication between a PC–based phone and a services that allow three or more endpoints to take part in a
phone on a traditional Switched Circuit Network (SCN). conference call. All terminals participating in the
Such requirements forced the need for a standard for IP conference establish a connection with the MCU. The
telephony. Version 2 of H.323, packet-based multimedia MCU manages conference resources, negotiates between
communications systems, was defined to accommodate terminals for the purpose of determining the audio or video
these additional requirements and was accepted in January coder/decoder (codec) to use, and may handle the media
1998. stream.
H.323 Version 3, approved on September 30, 1999, and
Version 4, approved on November 17, 2000, only makes
modest improvements to the Version 2 Recommendation,

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 IP Telephony: Technology and Standards

H.323 Protocols H.263 define methods for encoding analog video into
digital form for transmission and decoding back into analog
H.323 is a comprehensive and very complicated video code for replay. Because H.323 specifies support of
protocol. As shown in Table 2, it ties together a number of video as optional, the support of video codec is optional as
existing recommendations defined by International well.
Telecommunications Union (ITU) and Internet Engineering
Task Force (IETF). These protocols, together with some H.225 Call Signaling: H.225 call signaling is used to set
other recommendations, provide specifications for a range up connections between H.323 endpoints (terminals and
of communication including real-time voice, video and data gateways), over which the real-time data can be
transmission (see Figure 4). An overview of these protocols transported. Call signaling involves the exchange of H.225
is given below: protocol messages over a reliable call-signaling channel.
Table 2: H.323 Protocol Suite For example, H.225 protocol messages are carried over
TCP in an IP–based H.323 network. The call-signaling
Video Audio Data Transport channel is opened between two H.323 endpoints or
H.261 G.711 T.122 H.225 between an endpoint and the gatekeeper.
H.263 G.722 T.124 H.235
G.723.1 T.125 H.245 H.225 RAS: H.225 registration, admission, and status
G.728 T.126 H.450.1 (RAS) is the protocol between endpoints (terminals and
G.729 T.127 H.450.2 gateways) and gatekeepers. The RAS is used to perform
H.450.3 registration, admission control, bandwidth changes, status,
RTP and disengage procedures between endpoints and
X.224.0 gatekeepers. A RAS channel is used to exchange RAS
messages.
G.7xx Audio Codecs: The recommendations G.711 (audio
coding at 64 kbps), G.722 (64, 56, and 48 kbps), G.723.1 H.245 Control Signaling: H.245 control signaling is used
(5.3 and 6.3 kbps), G.728 (16 kbps), and G.729 (8 kbps) to exchange end-to-end control messages governing the
define the way in which analogue audio signals are operation of the H.323 endpoint. The messages carried
encoded into compressed digital form for transmission, and include messages to exchange capabilities of terminals and
decoded back into an uncompressed audio signal for replay. to open and close logical channels. The H.245 control
Because audio is the minimum service provided by the messages are carried over H.245 control channels.
H.323 standard, all H.323 terminals must have at least one
audio CODEC support. Q.931: Q.931 is a call signaling protocol and is used for
establishing H.323 calls. H.225 call control messages are
H.26x Video Codecs: The recommendations H.261 and embedded within the user-to-user elements of Q.931
messages to provide additional information not
available in Q.931 such as IP address
information.

RTP: Real-Time Transport Protocol (RTP)

provides end-to-end network transport
functions for applications transmitting real-
time data over IP networks. It provides
services such as payload type identification,
sequence numbering, time-stamping, and
delivery monitoring to real-time applications.
RTP can also be used with other transport
protocols. RTP is discussed in detail in an
earlier section.

RTCP: Real-time Transport Control Protocol

(RTCP) is the counterpart of RTP that
provides control services. The primary
function of RTCP is to provide feedback on
the quality of the media received using RTP.
RTCP is discussed in detail in an earlier
section.

Figure 4: H.323 Protocol Stack

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 IP Telephony: Technology and Standards

SESSION INITIATION
However, within peer-to-peer protocols, SIP is a much
more efficient and less complex protocol, therefore, more
PROTOCOL (SIP)
scalable than H.323. A high level SIP protocol stack is
shown in Figure 5.
The Session Initiation Protocol (SIP) emerged as an SIP Concepts
alternative to H.323, under the auspices of the Internet
Engineering Task Force (IETF). SIP is a much more Session: Session is the basic building block in SIP. A SIP
streamlined and powerful protocol, developed specifically session is a multimedia session consisting of a set of
for IP telephony. Smaller and more efficient than H.323, multimedia senders and receivers, and the data streams
SIP takes advantage of existing protocols to handle certain flowing from senders to receivers. All calls and
parts of the communication process. For example, Media conferences are established by setting up sessions among
Gateway Control Protocol (MGCP) is used by SIP to users.
establish a gateway connecting to the PSTN system. Conference: A conference is a multimedia session,
Since H.323 was originally designed for video identified by a common session description. A conference
conferencing over private, high speed LAN, it assumes can have zero or more members and includes the cases of a
details like authorization and user identification. H.323 also multicast conference, a full-mesh conference and a two-
provides all these extra services which are not usually party “phone call”, as well as combinations of these. Any
considered useful for a simple phone call. SIP, however, number of calls can be used to create a conference.
was designed from the beginning for multimedia sessions Call: A call consists of all participants in a conference
and conferences over Internet and wide area network invited by a common source. A SIP call is identified by a
(WAN). Because of these differences in their design globally unique call-ID.
objectives, SIP offers numerous compelling advantages in
the areas of extensibility, scalability, and ease of SIP Architecture Components
deployment over H.323. SIP specification defines a number of components that
SIP is transport layer independent. It can run over any are required to develop a SIP-based network. In many
datagram or stream protocol such as UDP, TCP, ATM, etc. implementations, some of these components are combined
It makes use of the Session Description Protocol (SDP) for into the same software modules.
specification of the session parameters. The audio or video
data streams are transported using RTP over UDP. SIP may SIP User Agents
use any IANA registered codec while H.323 requires ITU- A SIP user agent (UA) is a program that runs on a SIP
T defined standard only. device such as IP phones and gateways. It contains a client
SIP enables new services and applications not possible function and a server function.
with H.323 and other IP telephony protocols. For example, The user agent client (UAC) initiates SIP requests such
SIP uses a simple text-based encapsulation (based on the as initiating a call. It is the only entity on a SIP-based
Internet standard MIME) which enables it to transmit data network that is permitted to create an original request. The
and application programs with the voice call, making it UAC is also known as the calling user agent.
easy to send files, photos, and MP3 encoded information A user agent server (UAS) is one of many server types
during a call. It also enables developers to push the that receives SIP requests such as an incoming call and
intelligence to the edge of the networks, implement a sends back responses to those requests. A UAS is also
distributed architecture, and create advanced features. known as the called user agent. Normally, user agents are
Being peer-to-peer protocols, both SIP and H.323 discussed without any distinction made between their UAC
eliminate the need for central servers to control everything. and UAS components.
SIP servers
The SIP servers are distinguished by their roles played
by centralized hosts on a distributed network. There are
four types of SIP servers that can be implemented in a user
agent (UA). These are as follows:

Location Server: A location server is used to obtain

information about called party’s possible location. A
location is the IP address of the domain where a user is
located. To locate a user, the name of the user is sent to the
location server and the location server returns zero or
multiple locations (IP addresses of domains) where a called
party may be found.
Proxy Server: A proxy server is an intermediary program
that acts as both a server and a client for the purpose of
making requests on behalf of other clients. Requests are
Figure 5: SIP Protocol Stack

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 IP Telephony: Technology and Standards

serviced internally by a proxy server or transferred to other location of each communication endpoint and its media
servers. capabilities so that a level of service can be chosen that will
Redirect Server: A redirect server is a server that accepts a be possible for all participants. MGCP can be used to set
SIP request, maps the address into zero or more new up, maintain, and terminate calls between multiple
addresses and returns these addresses to the client. Unlike a endpoints.
Proxy, it cannot accept calls but can generate SIP responses The MGCP specifies a protocol at the Application layer
that instruct the UAC to contact another SIP entity. level that uses a master-slave model, where the gateways
Registrar Server: A registrar server is a server that accepts are expected to execute commands sent by the media
REGISTER requests. A client uses the REGISTER request gateway controllers. Two Media Gateway Controllers use
to let a proxy or redirect server know the location where the RTP to talk to one another and successfully transport voice
client can be reached. It provides a means whereby users packets.
can register their locations with a SIP server dynamically. MGCP is well suited for centralized systems that work
with dumb endpoints, such as analog phones. The most
MEDIA GATEWAY celebrated use of MGCP is for high-capacity gateways
designed to work with traditional telecom equipment.
CONTROL PROTOCOL
MEGACO / H.248
(MGCP)
Megaco/H.248 is the Media Gateway Control Protocol
Media Gateway Control Protocol (MGCP), endorsed by defined jointly by IETF and ITU-T for use in distributed
the Internet Engineering Task Force (IETF), is a protocol switching environments. The standard is endorsed by the
for handling the signaling and session management needed IETF as Megaco and by the ITU-T as Recommendation
during a multimedia conference. It is used for controlling H.248.
telephony gateways from external call control elements The Megaco/H.248 protocol was developed from the
called media gateway controllers or call agents. A Media Gateway Control Protocol (MGCP) – it refers to an
telephony gateway is a network element that provides enhanced version of MGCP. Megaco/H.248 provides
conversion between the audio signals carried on circuit- broadly equivalent functionality and has a very similar
switched network (such as PSTN) and data packets carried structure. The later Megaco/H.248 version supports more
over packet-switched networks (such as Internet). ports per gateway, as well as multiple gateways, and
MGCP assumes signaling control intelligence outside support for Time-Division Multiplexing (TDM) and
the gateways, in a media gateway controller (MGC). Asynchronous Transfer Mode (ATM) communication.
MGCP makes it possible for the MGC to determine the

VOICE QUALITY CHALLENGES AND SOLUTIONS

T raditional telephony networks are built to provide

an optimal service for time-sensitive voice
applications; these provide constant but low
bandwidth services. However, IP networks were built to
support non-real-time applications such as email, file
becomes unacceptable if the one-way delay becomes
greater than 450 ms. Delay is inherent in data networks
where it has no real impact, but with voice it is necessary
that Quality of Service (QoS) features are implemented
right across the IP network. Sources of delay in a packet
transfer and web information. These networks send and voice call include the collection of voice samples
receive data only as needed, in data packets. Because each (accumulation delay), encoding/decoding and packetizing
packet is individually routed across the network, this makes time, jitter buffer delays, and network transit delay. IP
packet-switching networks inherently less efficient in telephony gateways and terminals also contribute
dealing with voice traffic and poses a number of challenges significantly to delay.
to a quality voice transmission.
Jitter (Delay Variability): Jitter is the variation in inter-
CHALLENGES packet arrival time. In IP networks, packets that belong to
the same transmission do not always arrive with the same
It has been found that there are three factors that can amount of delay. This variation in delay is referred to as
profoundly impact the quality of the service. These are: “jitter.” The jitter causes gaps in the speech pattern; as a
result, voice transmissions may sound unnatural.
Delay (Latency): Delay in a voice network is the time
required for a voice signal to traverse the network. Two Packet Loss: IP networks do not provide guaranteed
problems that results from high end-to-end delay are delivery of the packets. The packets can be dropped under
“echo” and “talker-overlap”. Echo becomes annoying peak loads and during periods of congestion (caused, for
when the delay is more than 50 ms. It is perceived as a example, by link failures or inadequate capacity). For non-
significant quality problem. Talker-overlap can cause the real-time applications, such as email and file transfers,
receiver to start to talk before the sender is finished. It packet loss is not critical – the protocol allows

82 IEP-SAC Journal 2004-2005

 IP Telephony: Technology and Standards

retransmission to recover dropped packages. However, receiving device must interpret the lack of packets and re-
real-time voice data has to arrive within a certain time insert the silent spots, called comfort-noise, into the output.
window to be useful to reconstruct the voice signal.
Echo Cancellation: Echo cancellation is used to cancel the
SOLUTIONS echo caused by end-to-end delay of a voice transmission.
Echo cancellers monitor speech from the far end that passes
In order to deal with these issues and provide a voice through its receive-path and use this information to
service with a reasonable measure of quality, there are compute an estimate of the echo that is then subtracted
many techniques that are employed to deal with network from its send-path. ITU protocols G.165 and G.168 define
congestion and delay. These techniques include the the performance requirements that are currently required
following: for echo cancellers.

Prioritization: Prioritization is a method of guaranteeing Interpolation of Speech: Interpolation of speech is an

throughput for certain traffic on the network. This can approach used to compensate for packet loss. If certain
ensure that voice traffic on a data network is given high voice packets are delayed beyond a specific threshold, IP
priority. This prioritization can be based on location, telephony software interpolates by re-playing the last
protocol or application type. RSVP is designed to ensure packet, and sending of redundant information. In order to
this QoS. help ensure a quality voice conversation, packet losses
greater than 10% are not tolerable. For packet losses under
Fragmentation: Fragmentation divides the packets into 10%, interpolation can help maintain a continuous flow of
smaller fragments so that their priority can be ensured. This voice with minimal distraction to the quality.
can help reduce the overall delay of voice delivery.
However, this can create extra overhead because of the Speech Codecs: A speech codec transforms analog voice
large size of IP headers (20 bytes). So although necessary, into digital bit-streams, and vice versa. In addition, some
fragmentation alone cannot ensure the reliable delivery of speech codecs also use compression techniques to reduce
real-time voice applications. the transmission bandwidth required. Most PSTN networks
use ITU-T G.711 recommendation that encodes the speech
Jitter Buffering: Jitter buffering is a technique that allows at 64 kbps. Current ITU-T recommendations include
packets to be collected into a buffer and held there long codecs that compress to as low as 5.3 kbps, although
enough for the slower packets to arrive so that they can all quality at this bit rate is well below that of the highest
be played in proper sequence. Although this can remove quality G.711. Essentially, compression is a balancing act
packet delay, this creates additional overall delay. The best between voice quality, local computation power, delay, and
compromise is to fit jitter buffer in the network's network bandwidth required. In general, the lower the bit
differential delay. This will provide the necessary balance rate, the lower the quality perceived by the listener.
between the packet delay and the overall delay. However, more modern codec designs are driving up the
quality for a given bit rate. The more recent, though
Silence Suppression: Silence suppression is a technique computationally intensive codecs, GSMEFR at 12.2 kbps
that is used to suppress the transmission of silence. These and G.728 at 16 kbps, are comparable to the low-
techniques take advantage of half-duplex nature of human complexity G.726 codec that offers good performance only
conversation (one person listens while the other talks) by at 32 kbps or above.
detecting when there is a gap and then suppresses the
transmission of these silences. This can amount to 50-60% In order for different manufacturers to implement these
of the time of a call, resulting in considerable bandwidth various techniques and maintain interoperability, the
conservation. However, because these silences are standards like H.323, SGCP, SAP, SIP, RTSP and SDP
necessary for the conversation to sound natural, the have been recommended and approved.

CONCLUSION

D
networks.
ata traffic has traditionally been forced to fit onto
the voice network. The Internet has created an
opportunity to reverse this strategy – voice, video,
fax and other multimedia can now be carried over IP
quality isn't as good as conventional phone calls, the
discounted prices make up for it.
Although the price of IP call is now negligible, the
phone companies will be likely to object to the free long
distance service offered by the Internet and may raise the
IP telephony users enjoy free long distance phone calls, price of local phone calls in response. It remains to be seen
coupled with numerous additional features. Both traditional whether or not IP telephone calls will continue to be such a
telephone features, such as call-waiting and voice mail are good bargain to the average user, since the pricing for voice
included, as well as non-traditional features, such as group traffic is now undergoing change.
chats and text-based document sharing. Although the

IEP-SAC Journal 2004-2005 83

 IP Telephony: Technology and Standards

Voice communications will certainly remain a basic unifying agent, regardless of the underlying architecture
form of interaction for all of us. The public switched (e.g., leased lines, frame relay, or ATM) of an
telephone network simply cannot be replaced, or even organization's network.
dramatically changed, in the short term. The immediate The market for IP telephony products is established and
goal for IP telephony developers is to reproduce existing is beginning its rapid growth phase. The use of off-the-shelf
telephone capabilities at a significantly lower “total cost” software and hardware components can allow for a rapid
and to offer a technically competitive alternative to the implementation and a great degree of flexibility in the
PSTN. implementation.
As data traffic continues to increase and surpass that of The Internet and its underlying IP protocol suite have
voice traffic, the convergence and integration of these become the driving force for new technologies. Future
technologies will not only continue to improve, but will extensions will include innovative new solutions including
also pave the way for a truly unified and seamless means of conference bridging, voice and data synchronization,
communication. The economics of placing all traffic combined real-time and message-based services, text-to-
(voice, video and data) over an IP network will pull speech conversion and voice response systems.
companies in this direction, simply because IP will act as a

*** Date: 8th October 2004

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84 IEP-SAC Journal 2004-2005