Parched For Services?
Here, Try A SIP
BY JONATHAN ROSENBERG
Anyone involved in the Voice over IP (VoIP) industry over the last few
years has seen that there have been many twists and turns on the road to
deployment. New protocols, such as MGCP (Media Gateway Control Protocol)
and SIP (Session Initiation Protocol), have entered the arena and caused
changes in directions. New architectures, such as the softswitch, have
challenged service providers to think carefully about how interfaces to
the traditional SS7 networks will be handled. Many industry observers
believe another major shift lies ahead: the emergence of true converged
applications that will integrate the "brains and brawn" -- the
intelligent architecture and carrier-grade versatility and scalability --
that service providers of all sizes will need.
Service providers and vendors have certainly talked about services and
features, but so far these have been slow in coming. The primary
applications of VoIP to date have been toll bypass and PC-to-phone calling
-- all with the end goal of "cheap long distance." But the price
differential of VoIP may be short-lived.
One need only watch the media to see the constant barrage of
commercials advertising lower and lower prices for long distance calls.
Within the United States, long distance calling has gone from
distance-sensitive to distance-insensitive, and has dropped in cost from
20 cents to 10, to 7, and now as low as 5 cents a minute. Wireless
providers are already experimenting with flat-rate pricing, whereby calls
to a limited set of phone numbers are flat rate. A long-distance provider
in Canada is already providing flat-rate service anywhere within the
country. International calling has seen a similar decline in rates.
Given, then, that IP telephony is not likely to be much (if at all)
cheaper than traditional telephony, and given that it will not have the
same maturity and features users expect from traditional telephony for
some time, why should they accept it?
The answer lies in the advent of true converged services. This means
not simply porting traditional features like VPN and Centrex to IP. True
converged features are about integrating Web, e-mail, presence, chat, and
instant messaging with voice. This integration allows these other IP
applications to become integral components of new voice services. These
kinds of services, fundamentally, cannot be done in the circuit-switched
world. They provide something new, something different -- something of
value for service providers and their customers.
Converged services and applications also open the door to service
provider differentiation. How? By taking advantage of what we may call
"Rosenberg's Law." This theorem holds that the set of features
and services that a service provider can deliver increases exponentially
with the number of applications (such as Web, e-mail, and voice) that are
combined to provide them. For example, take transfer, add Web, and you've
got two service variants. Add e-mail, and you've got four. With such a
large set of services, it's a fertile ground for differentiation.
What, specifically, will these converged services look like? Consider a
few examples.
Web IVR
Interactive Voice Response (IVR) systems were supposed to make it easier
for consumers to get the information they need from businesses. However,
for most people, they are a nightmare to navigate and can often be
frustrating. No two IVR systems use the same prompts or have the same
structure. Consumers often have trouble remembering which option is the
one they want, and have difficulty figuring out how to reach an actual
human being.
Not surprisingly, the Web is a much more natural means for navigating
these systems. Instead of pressing buttons on phones, users can click on
links. Going back is as simple as hitting the back button on the browser.
Users can quickly scan the available links at any level to find the right
place for them.
The idea behind Web IVR is to combine the rapid communications of voice
with the ease of navigation of the Web. Using their IP phones, users can
call a number. During the day, they might be directly connected to a
person. After business hours, they can be redirected to a Web page that
contains a Web version of the IVR system. As users click through the
system, they may eventually reach a point where a click actually completes
the phone call to a real person.
Multimedia Caller ID
While caller ID is a popular consumer service, it still provides only
limited information about the caller: phone number, and occasionally,
name. The data is usually presented in a simple one- or two-line LCD
display. What if, instead, a wide range of multimedia were be used to
identify the caller? The multimedia could include text, audio, images, and
even videos. We refer to this enhanced caller ID service as multimedia
caller ID.
The most compelling example is a simple thumbnail photo of the caller.
This would be ideal for people who are good at remembering faces, but not
names. When someone calls, both his or her picture, along with the name,
would appear. Another useful piece of information is a small audio snippet
containing the caller's spoken name. Instead of just ringing, the phone
might actually ring and then deliver an audio message such as "call
from John Smith" in the actual voice of the caller.
Presence Notification Of Availability
The simple telephone has served as an ideal tool for real-time
communications between people. But for the purposes of leaving messages,
many users prefer e-mail, since it is more concise and is more easily
stored and manipulated. For these users, a call that ends in voice mail is
a useless call. To solve this problem, traditional buddy lists and
presence systems can be combined with voice.
Buddy lists allow a user to subscribe to a list of friends. When one of
those friends comes online, the user is notified, and they then have the
option of sending an instant message. When coupled with voice, the service
becomes very compelling. Rather than presence being a simple
"on" or "off," indicating the ability to receive
instant messages, it can reflect a person's readiness and ability to
communicate using a variety of different means. For example, a user can
subscribe to the cell phone of a second user. When the second user turns
his or her cell phone on, the first user gets a notification, and can then
send e-mail.
Another useful variation on this service is to allow one user to be
notified of status changes on another user's traditional telephone. This
service is used often in PBXs to allow an assistant to know when the boss
is on or off the phone. With the Internet, this service can be made
global. Even when the boss may be in another country, the assistant can
subscribe to his or her phone to gauge availability.
WHY SIP?
The Session Initiation Protocol (SIP), specified in IETF RFC2543, is a
powerful tool for call control and signaling that is gaining tremendous
support among service providers and vendors. SIP turns out to be an ideal
protocol for providing truly converged applications. This is primarily
because it borrows so heavily from other Internet protocols, and in
particular, HTTP and SMTP. Three features render SIP ideal for converged
services:
- MIME. MIME stands for Multipurpose Internet Mail
Extensions. MIME is now used in a number of other protocols, including
HTTP. It has become the de facto standard for describing all sorts of
content on the Internet. Every audio format, every video type, and
every image is a registered MIME type. SIP uses MIME, and as a result,
SIP messages can contain Java applets, images (for multimedia caller
ID), audio files (to announce the name of the caller), authorization
tokens, or even billing data.
- URLs (Universal Resource Locators). URLs have become
the Internet standard way of addressing. Web pages are identified by
HTTP URLs. E-mail is sent by clicking on "mail to" URLs. As
new applications are developed, they define URL types as well. As a
result, URLs have become the universal way to access Internet
applications. Rather than defining some new type of address space, SIP
uses URLs, and only URLs, for addressing. This means that it is just
as easy to redirect someone to another phone as it is to redirect
someone to a Web page. Transferring to a PC voice terminal is the same
as transferring to email.
- DNS. The Domain Name System (DNS) is the global
directory service used on the Internet for translating domain names
(such as www.yahoo.com) into IP
addresses. The DNS is the key component needed to deliver Web and
e-mail requests to the server that can appropriately handle them.
Rather than inventing new mechanisms to accomplish this for VoIP
calls, SIP borrows from the e-mail model. It uses DNS in exactly the
same way e-mail routing makes use of DNS. The substantive benefit of
this approach is that companies can share their e-mail routing
infrastructure for voice call routing as well. This also simplifies
integration of voice and e-mail. Servers along the call path can
easily create and forward e-mail messages, and vice versa, enabling
various combined services.
A PLATFORM FOR DELIVERING APPLICATIONS
Given that these converged services are the key to revenue for
Internet telephony, what does a service provider need to deliver them? The
answer: a converged applications server. This server is contacted by
softswitches, PCs, and gateways when an application needs to be invoked
for a call. In order to deliver converged services, such a platform must
support not simply voice, but e-mail, Web, instant messaging, and
presence. Its service creation tools must mirror those used on the Web,
rather than those used in the IN (Intelligent Network) world. With such a
server in place, the opportunities for service providers to deliver new
services -- and the revenues associated with them -- are virtually
limitless.
Jonathan Rosenberg is chief scientist for dynamicsoft. For more
information, visit www.dynamicsoft.com.
Also, Rosenberg co-chairs the Internet Engineering Task Force's (IETF) SIP
working group, and chairs the IETF's IP Telephony working group. |