From where does IP telephony derive its power? New applications. Some of
these applications bring the human touch to e-Business. Others serve the
collaboration needs of an increasingly distributed workforce. And yet
others signal opportunities for enhanced connectivity, specifically, a
broader array of connectivity options that take advantage of Internet
ubiquity and LAN plug-and-play capabilities. However, if these
applications are to work -- if they are to support IP telephony's
ascendance -- they must have the benefit of an environment characterized
by interoperability. That is, applications will rely on interoperability
among IP telephony systems, as well as on interoperability between the
worldwide public and private telephony networks. And ultimately, such
interoperability will rely on IP telephony protocol standards.
Accordingly, these standards are no less important today than telephony
standards have been over the last hundred years.
IP telephony protocols constitute a framework, and the framework should
accommodate the essential functional components of IP telephony systems.
These components include:
- IP telephones (and appliances) and PC software telephony clients.
- Gateways to the circuit-switched world and to legacy analog and
digital phones.
- Application servers.
- Communications servers (also known as call server and connection
managers).
Ideally, these components should be tied together by IP telephony
protocols. But exactly how should these components be tied together? In
other words, what will the architecture look like? This question has
occasioned divergent (or seemingly divergent) responses.
The main differences arise over how best to distribute intelligence to
implement scalable IP telephony systems. Some approaches emphasize thin IP
telephony clients, which rely primarily on network intelligence to
initiate and manage communications. Other approaches emphasize thick IP
telephony clients, which rely primarily on end point intelligence.
Ultimately, both architectural approaches will co-exist in real-world
systems.
RECONCILABLE DIFFERENCES?
The discussion over thin and thick IP telephony clients revisits the
discussion over smart and dumb networks. It goes something like this: Do
you make the end points relatively dumb (the telephony model), and put the
intelligence in the network? Or do you put all the intelligence in the end
points (the PC model), and use the network as a dumb conduit? The debate
is around the question of what's the right architecture to deliver feature
functionality in the form of IP telephones to end-users with the right
price points.
Thin clients use what is called "stimulus signaling," working
into communications servers that provide most of the feature
functionality. In this case, if a feature key such as three-way
conferencing is pressed, then this action is signaled to the
communications server, which then makes it all happen. Thick clients, on
the other hand, use "functional signaling," minimizing reliance
on the communications server, which provides address translation or
routing functions.
Thin client proponents argue that their approach allows overall system
cost optimization, as well as ease of horizontal services development and
deployment. Thick client proponents argue that memory and processing are
cheap, and that their way is the right way to build IP telephony systems.
In either case, the ability to negotiate coding schemes (including
wideband coding for high-quality audio) is provided to ensure audio
compatibility between the communicating devices.
Interesting debates! In reality, both client types will be available,
and the user will decide which offers superior feature functionality,
operational ease, and cost. The good news is that the thin client approach
is entirely compatible at the system level with the thick client approach,
with interworking provided by the communications server. In fact, think of
the communications servers as providing thick client functionality for a
set of thin clients. In addition, everyone agrees that thick clients will
exist, for example, in PCs and application servers, and that the protocols
being developed for thick clients will also apply to communications
between communications servers.
PROTOCOLS THICK: H.323 VIES WITH SIP
If you were to start with a clean slate, you might find it daunting to
define protocols that would operate in function-rich environments. You
would confront many difficult protocol questions. For example, should
signaling be defined for each feature to be supported (for example,
setting up a conference call) or should more primitive signals be defined
(for example, adding a user to a call)? And which might you then combine
in various ways to define new modes of operation? What coding scheme
should be used to define the protocol syntax? What are the precise
meanings of each signaling element that define the protocol semantics?
Should protocols be developed that are optimized for IP telephony with
many short real-time interactions and a human at the end of the line? Or
should IP telephony leverage existing protocols and emphasize re-use
rather than optimization?
As has often happened in the complex world of telecommunications
protocols, two camps have evolved. The H.323 camp sees IP telephony as an
element of the more general multimedia conferencing space and argues that
there is no need to re-invent the wheel. The SIP camp says that H.323 is
too complex and that a new IP telephony protocol is warranted that fully
exploits the dynamic nature of IP networking.
H.323 is part of a broader family of standards developed by the
International Telecommunication Union (ITU), the body that is responsible
for most telecommunications protocols -- with the conspicuous exception of
IP. This family includes the widely deployed H.320 standard, which is used
in virtually every room video conferencing system and runs over the public
ISDN network. H.323, however, defines an IP-based video conferencing
architecture. In addition, it relies on PC intelligence and on a
communications server (called a gatekeeper in the standard) for
conferencing coordination. It is widely available in PC software (for
example, Microsoft NetMeeting). However, H.323 hasn't been widely used for
a several reasons, including limited business value, lack of end-user
familiarity, and impacts on the network. In short, it hasn't been an IT
priority to date.
SIP, the Signaling Internetworking Protocol, is a new standard being
developed by the Internet Engineering Task Force (IETF), which is the de
facto center of IP standardization. It takes the classical IP networking
approach of allowing two end points to establish communication directly,
fully leveraging standard IP capabilities such as DNS (Data Name Server)
and mobile IP services. Extensions are now being developed to provide IP
telephony features in a similar manner.
In the short term, H.323 is the de facto interoperability standard.
However, SIP continues to mature in terms of features/functionality. It is
likely that SIP will eventually displace H.323 as the preferred IP
telephony thick client and inter-communications server signaling protocol.
The long-term success of H.323 will be tied to multimedia conferencing.
Within enterprise networks, both types of clients will need to be
supported with interworking provided via communications servers,
particularly in the early days of IP telephony. SIP will be with us for a
long time, while H.323's survival will be determined off the battlefield.
PROTOCOLS THIN: CONVERGENCE VIA MEGACO AND H.248
Defining a thin client IP telephony protocol is conceptually a simpler
task. While several protocols have been proposed, even in the relatively
few years since the emergence of IP telephony, standardization of a single
protocol is at hand. The effort is being carried out jointly by the IETF
Media Gateway Control Working Group (Megaco) and the ITU. The IETF group
calls the result of the joint effort the Megaco protocol, and the ITU
calls it H.248. However, the Megaco protocol and H.248 are synonymous.
While the work originated in the IETF, the ITU will most likely control
the actual documentation and will evolve the standard towards richer
multimedia support.
The Megaco/H.248 protocol provides a flexible framework for
master/slave control of a very broad range of IP telephones and similar
devices. It lends itself well to definition of user interface elements
such as function keys, indicators, displays, and so on. This capability,
perhaps best expressed in the creation of protocol profiles, makes it
possible to define very simple devices, greatly reducing complexity/cost
and improving interoperability. The Megaco/H.248 protocol can apply to
applications such as PSTN trunking gateways, ATM interfaces, analog line
and telephone interfaces, and announcement servers.
The relationship between the H.248/Megaco protocol and MGCP (Media
Gateway Control Protocol) has been an area of some confusion.
Historically, MGCP was to have been the acronym used for the output of the
Megaco working group. However, the term was also used by Telcordia and
others as the title of their technical draft proposal targeted at basic
"black" telephones and gateways to the PSTN. That proposal was
not accepted by the Megaco working group, although many important aspects
were incorporated along with inputs from several other proposals.
Unfortunately, Telcordia published their MGCP proposal as an IETF
informational RFC and continued to promote MGCP as a de facto standard.
SOME OTHER STANDARDS PIECES
Additional standards pieces merit attention. One such standards piece,
developed by a working group within the Telecommunications Industry
Association (TIA), concentrates on VoIP telephones. The group means to
define a "whole device" standard covering all the considerations
that would arise during the manufacture of a business-class IP telephone.
These considerations include everything from Ethernet physical
interfacing, to audio quality requirements, to safety, and everything in
between. This specification is largely made up of references to other
standards, and includes three options for a control protocol: Megaco IP
Phone, SIP, and H.323.
Another TIA working group is concentrating on audio quality standards.
In addition, an IEEE committee is working on Ethernet powering.
THICK OR THIN?
We can expect both thick and thin clients in the network. Thin clients
leverage intelligence in the network, facilitating the introduction of new
features (with no need to download software to potentially thousands of
devices) and lowering the total cost of ownership. Thick clients will be
used in more sophisticated PC-based environments, such as in customer
interaction centers, and as the base of signaling between communications
servers. In either cases, buyers should understand the fit of any
particular IP telephony environment to their business needs, keep an eye
on total cost of ownership, and ensure that stronger interworking is
provided with their installed base.
Tony Rybczynski is director of strategic marketing and technologies
for Nortel Networks' Enterprise Solutions unit. For more information,
visit the company's Web site at nortelnetworks.com.
E-mail questions or comments to [email protected].
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