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NBX Information
An Overview: What is LAN Telephony?
Total Solutions
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©2002 Total Technologies, All Rights Reserved Some Stories Courtesy 3Com Corporation
LAN
Telephony
An Overview of Market Drivers, Applications,
and Technology
Introduction
LAN telephony, which means "the integration of telephony and data services
provided by packet-switched data networks," is not just a new high-tech
buzz term. It is the technology that takes person-to-person communication to a
high new level and associated costs to a lower level. LAN telephony enables the
true integration of e-mail and v-mail, and provides easier, more flexible, and
cost-efficient use of many other applications, such as automated call
distribution, interactive voice response, voice logging, etc. This is in
contrast with the relatively limited integration offered by the current
voice/data integration paradigm, computer-telephony integration (CTI), in which
voice traffic is kept separate from data traffic and carried over
circuit-switched links, and telephone services are enhanced by computer support.
One of LAN telephony's most intriguing values is that it enables substantial cost savings on long-distance voice calls, data transmissions, and video-conferencing. Another important value of LAN telephony is the productivity improvement that it offers through new, integrated voice and data applications.
LAN telephony is not relegated only to LANs, by the way. Such a service may also involve wide-area data networks; in some cases, it may not involve a local area component at all. One of the most common uses of LAN telephony will be in the enterprise intranet environment, referred to as IP telephony. Note that this service should not be confused with Internet telephony, which is based on hobbyist-type, low-cost software solutions making use of the public Internet. Internet telephony offers rather low quality compared to today's toll-quality voice, as a result of the limited quality that the software codecs can offer, as well as the lack of service guarantees on the Internet. In contrast, IP telephony, with its toll-quality, hardware-based codecs and its use of the well-engineered enterprise intranet, is poised to offer quality comparable to the plain old telephone service (POTS).
Because of these advantages, there is a great deal of interest in the market for LAN telephony devices. This report:
Examines the key factors driving the market
Evaluates current LAN telephony technologies and their applications and identifies the places where new technologies are needed
Examines the new types of applications that can be enabled by those new technologies
LAN Telephony Market
Drivers and Product Vendors
An important driving force behind LAN telephony is cost savings, especially for
corporations with large data networks. The high cost of long-distance and
international voice calls—thanks to layers of local and international
carriers—is the crux of the issue. A significant portion of this cost
originates from regulatory taxes imposed on long-distance voice calls. Such
surcharges are not applicable to long-distance circuits carrying data traffic;
thus, for a given bandwidth, it is much less expensive to make a data call than
a voice call.
In addition to the cost savings for long-distance voice calls, carrying voice traffic on the data network within a business building or campus also can achieve substantial cost savings, since the operation of today's proprietary PBX setups is relatively cost-inefficient.
There are other significant motivating factors for carrying voice traffic over data networks. An important benefit of LAN telephony is the integration of voice and data applications, which can result in more effective business processes. Examples of such applications are integrated voice mail and e-mail, teleconferencing, computer-supported collaborative work, and automated and intelligent call distribution.
While these factors have always existed to motivate LAN telephony, the technology only recently became feasible as a result of a number of other contributing factorse:
The trend of deregulation of telecommunications in the United States as well as other countries has promoted a competitive environment of network service providers (NSPs) to proliferate. Many smaller NSPs and telecom corporations, trying to carve out a niche for themselves, have begun to actively develop LAN telephony.
The explosive growth of the Internet and the World Wide Web (WWW), leading to connectivity that will eventually approach that of telephone networks.
The de facto standardization of the personal computer on "Wintel" architecture has helped create an environment in which compatibility and interoperability issues are less significant. This is also connected with Intel's planned obsolescence strategy, leading to faster and faster processors every year, which are now easily capable of compressing/decompressing audio in software with a large fraction of processing power to spare for other tasks.
As a result of these factors, the market today is ready for LAN telephony. For example, in Figure 1 we show the high rate of growth projected for Internet telephone service in the coming years. In the long term, Web commerce and productivity improvement will drive integrated services even more over data networks, with broadband access technologies such as xDSL and cable modems providing a low-cost vehicle for integrated services. The result will be transparent telephony services across circuit and packet-switched networks.
Having recognized the tremendous growth potential in the LAN telephony market, a number of companies are pushing the market from several directions:
Cisco Systems and many networking startups are in the market for toll-bypass systems and key system/PBX replacements.
Intel, in addition to software giants Microsoft and Netscape, are promoting voice over the Internet and voice capabilities within the Web.
CTI vendors want to provide greater value by tighter integration of voice and data via LAN telephony.
PBX and current telephony users are seeking ways to reduce costs.
Carriers and NSPs are looking for ways to better leverage their data networking infrastructure.
Telecom and PBX equipment vendors are also engaged in the LAN telephony market, but they are reluctant to cannibalize their current sales.
Figure 1: Internet Phone Projections
State of the Art:
Computer Telephony Integration (CTI)
The old paradigm for integrating data and voice has been to use the
circuit-switched telephony fabric for data communications. This has obvious
drawbacks, such as the relatively low bandwidth available to data traffic, the
inefficiency of circuit-switched data communications due to the "bursty"
nature of data traffic and the limited voice/data integration possibilities
offered by this paradigm.
The current standard is in a transitory stage between the old paradigm and the full LAN telephony paradigm, in which a loose integration between circuit and packet-switched networks is provided and voice is carried by a circuit-switched network.
Figure 2: CBX: The Original CTI System
The CTI concept was introduced about a decade ago with a product called CBX (Figure 2). The CBX consisted of a PBX, which contained a CPU in addition to the usual line and trunk interfaces: the PBX is attached to an attendant console through a proprietary interface. As personal computers and LANs became more commonplace, the PBX CPU and proprietary console were replaced by one or more server computers interconnected by a LAN.
Custom CTI servers are usually bundled with proprietary PBXs or voice circuit switches. A typical contemporary CTI system setup is shown in Figure 3. One major application of this example setup is integrated v-mail and e-mail. In a basic form of that application, only the v-mail envelope information is sent to the user via e-mail, and the v-mail itself is played back via the user's telephone. In a more advanced form, v-mail can be sent as a digitized voice clip enclosed in the e-mail message, which can then be played back on the user's PC.
A similar arrangement can also be used for voice logging, which is used mainly by financial and emergency-response organizations.
Other applications of CTI include:
Automated call distribution (ACD), whereby an ACD server performs value-based queuing of incoming calls (Figure 4)
Teleconferencing, in which normally DSP-based custom conference servers are employed and users join the conference using the DTMF tones (Figure 5)
Interactive voice response, in which responses are preprogrammed in a server as a workflow component (Figure 6)
Figure 3: A Typical CTI System for Integrated E-Mail/V-Mail and Voice Logging
Figure 4: A CTI Setup for Automated Call Distribution (ACD)
Figure 5: A CTI Setup for Teleconferencing
Figure 6: A CTI Setup for Interactive Voice Response (IVR)
While the CTI paradigm enables many enhancements to the telephony applications, it does not bring the cost savings that can be associated with true LAN telephony; this again is due to expensive proprietary PBX equipment. Furthermore, since voice calls are carried separately from the data network, it becomes difficult, if not impossible, to realize many integrated applications.
LAN Telephony
Applications
An important group of LAN telephony users will be mobile users. With LAN
telephony, users working away from their offices (i.e., at home or in a hotel)
can use a single phone line to carry both data and voice traffic. Users would
dial up to directly access their intranet, which would be engineered to carry
real-time audio traffic. Such a system also provides an integrated directory
view, enabling remote users to locate individuals within the corporation for
voice- or e-mail connection in a unified way. Likewise, phone callers (internal
or external to the corporation) can locate the mobile workers connected to any
part of the intranet. Thus, LAN telephony allows users to work seamlessly from
any location (Figure 7).
Figure 7: Seamless, Integrated Voice/Data Access via LAN Telephony
Teleconferencing and computer-supported collaborative work applications would also be greatly enhanced by carrying voice and video streams over the corporate intranet as illustrated in Figure 8. By using the LAN-based conferencing standards, transparent connectivity of different terminal equipment can be achieved; the media used by any conference participant would be limited only by what is supported by his or her terminal equipment. Connectivity to room-based conference systems or analog telephone users would be achieved by means of gateways, which would perform the required protocol and media translations.
Figure 8: LAN Telephony Enhances Teleconferencing and Computer-Supported
Collaborative Work
LAN telephony would also allow true integration of e-mail and v-mail, where v-mail can be delivered as e-mail to be played back at the user's PC, or e-mail can be retrieved as v-mail and read to the user over the telephone by a synthetic voice. Where media conversion capability is inadequate (such as image-to-speech), at least the envelope information can be made available. A typical scenario involving LAN telephony–based unified e-mail/v-mail is shown in Figure 9.
Figure 9: LAN Telephony Allows Truly Unified E-Mail and V-Mail Messaging
Voice logging also becomes easier and more flexible with LAN telephony, because any voice stream can be logged, and the stored voice clips can be played back from any terminal connected to the network. (See Figure 10 for an example scenario.) Password protection and encryption can be applied for security purposes.
Figure 10: Voice Logging Becomes Easy and Flexible with LAN Telephony
LAN Telephony
Technologies
Considering the ISO/OSI network model, specific functions must be provided in
each layer to support LAN telephony. At the application layer,
application-specific policies need to be specified and their execution
coordinated. Presentation and session layers need to handle the session
coordination (where each session might include hundreds of participants
listening in a conference): and directory lookup services. Relevant protocols in
those layers are RTSP, RTP, LDAP, H.225.0, H.245, Q.931, IP/E.164 address
translation, etc.
Figure 11: Functions Needed in Each Layer to Support LAN Telephony
At the transport and network layers, mechanisms must exist to provide a good quality of service for audio, by means of the RSVP and ISSLL specifications, as well as appropriate queuing/scheduling mechanisms, such as weighted fair queuing (WFQ), cost-based scheduling, etc. It is the data link (especially MAC) layer's responsibility to provide predictable delivery services; 3Com's PACE™ technology, as well as 802.1p, 802.3x, LANE 2.0, etc., are applicable here.
At the physical layer, the main issue is bandwidth. While an individual audio stream occupies relatively small amounts of bandwidth (i.e., 8 to 64 Kbps) compared to the bandwidth of a LAN, it is still important to provide sufficient access bandwidth to a remote user to comfortably have good-quality audio in addition to data access at a decent rate. Even in the corporate workgroup or backbone environments, the multitude of voice calls can place a high demand on the required bandwidth.
The standards groups that work on LAN telephony–related technology include:
ITU-T, which develops standards for teleconferencing systems and protocols, as well as for audio/video encoding
International Multimedia Teleconferencing Consortium (IMTC), and in particular its Voice over Internet Protocol (VoIP) Forum, which produces interoperability agreements within the industry (which are H.323-compliant)
European Telecommunications Standards Institute (ETSI)
Internet Engineering Task Force (IETF), which has working groups on RTP, RSVP, ISSLL, etc.
IEEE 802 (in particular, 802.1p/Q, and 802.3x)
In the following sections, we first analyze the most important building block of LAN telephony, the ITU-T H.323 standard, which specifies the visual telephone system and equipment for packet-switched networks. We then discuss the remaining technology issues and challenges for LAN telephony and describe the areas where technology development would achieve significant value.
ITU-T H.323
H.323 is an umbrella standard that covers various audio and video encoding
standards, H.225.0 packetization (based on IETF's RTP specification) and call
control (based on the ITU-T Q.931 signaling protocol), as well as the H.245
protocol for capability exchange between terminals. Figure 12 illustrates the
scope and architecture of an H.323 terminal.
Figure 12: Scope and Architecture of H.323 Terminal
In Figure 13, a typical software architecture for a H.323 system is shown for a Windows PC connected to an IP network. On the sending side, the uncompressed audio/video information is passed to the encoders by the drivers, then given to the audio/video application program. For transmission, the information is passed to the terminal management application (which may be the same as the audio/video application); the media streams are carried over RTP/UDP, and the call control is performed using H.225-H.245/TCP.
Figure 13: H.323 Software Architecture on a PC Connected to an IP Network
Gateways provide the interoperability between H.323 and the public-switch telephone network as well as networks running other teleconferencing standards such as H.320 for ISDN, H.324 for voice, and H.310/H.321 for ATM (Figure 14). An example H.323 deployment scenario is shown in Figure 15. H.323 terminals in the same local area are interconnected by a switched LAN. Access to remote sites is provided by gateways, routers, or integrated gateway/router devices. The gateways provide communication with H.320 and H.324 terminals remotely connected to the ISDN and PSTN, respectively. H.323-to-H.323 communication between two remote sites can be achieved using routers that directly carry IP traffic over PPP running on ISDN. For better channel efficiency, gateways can translate H.323 streams into H.320 to be carried over ISDN lines, and vice versa.
Figure 14: Interoperability of H.32x Terminals
Figure 15: Example of H.323 Deployment Scenario
Call processing, address translation, and distributed application manager functions are coordinated by H.323 gatekeepers. An example scenario is shown in Figure 16. An analog phone call made into an H.323-based response center is received by the gatekeeper, which checks with an advanced call management application to determine which agent to forward the call to. In this case, all agents turn out to be busy, so the call is forwarded to a unified message server, where the customer leaves a voice-mail message. The unified message server then sends a notification of the v-mail to an agent application, which redirects it to an information fulfillment workflow server, which in turn initiates workflow. All these activities are coordinated by the gatekeeper.
Figure 16: Example Scenario of Distributed Application Management by H.323
Gatekeeper
Technology Issues and Challenges
Transport Issues
In order to carry voice modem lines along with data and/or video, the voice has
to be compressed down to, at most, 8 Kbps. The standard compression techniques
used today for such data rates (i.e., G.723.1, G.7297) introduce sufficient
distortion that the DTMF tones cannot be carried as audio signals and be
expected to be correctly decoded by the receiver. Therefore, they need to be
separated from the audio signal at the sender and conveyed separately to the
receiver. Two approaches for carrying the DTMF information are in-band via RTP
or out-of-band via H.245. Each has advantages and disadvantages, and the VoIP
Forum has chosen to recommend that either approach be used to convey DTMF.
Another transport issue is to provide sufficiently low end-to-end delay and jitter by putting QoS request/negotiation/enforcement mechanisms in place. At the network layer, RSVP and ISSLL specifications are building blocks toward that end. In addition, QoS-based routing mechanisms are being developed, and PPP is being extended to provide priorities for different classes of traffic.
Interoperability Issues
The diversity of network types is something that will continue to exist in the
foreseeable future. Translations between many signaling mechanisms need to be
provided, such as that between SS7/Q.931/PBX (proprietary) call setup signals,
framing conversions between H.221 and H.225.0, and capability exchange protocol
conversions between H.242 and H.245. Likewise, IP address, user name and E.164
number translations need to be provided by supporting a closer integration of
DNS, DHCP, and LDAP services compared to what exists today. For example,
currently one cannot obtain the address of the DNS server automatically during
the DHCP initialization. On the network management side, SNMP MIBs need to be
developed to support various protocol and application entities; also,
appropriate policy specifications must be defined, and mechanisms for correctly
interpreting them must be deployed.
API Issues
The main issue today with the telephony APIs is the existence of multiple APIs,
such as TAPI 2.x, JTAPI, and ECTF S.100/200. This makes it difficult for
application developers to come up with truly platform-independent, portable
applications.
Technology Developments
For terminals, the important technologies and products to be developed are:
Ethernet phone
Voice-capable information appliances
Integrated telephone/set-top boxes
Intelligent agents
For gateways, the technologies of importance are:
Scalable, distributed DSP architectures
ASICs with DSP core
Audio/video codes such as G.723.1, G.729A, H.263, H.262 (MPEG 2)
Transcoding of audio and video
DTMF processing and signaling
Digital audio mixing
In the IP telephony infrastructure, technology developments can be divided into three phases as illustrated in Table 1. During phase 1, basic gatekeepers would be deployed to perform call processing (such as connection), supplementary services (such as call forwarding/waiting/holding), IP/E.164 address translation, dial plan (which can be considered a simple form of policy specification), accounting, and an API to Advanced Call Management. As for directory services, gatekeeper/DNS integration can be achieved during this phase. The gateway is incorporated as part of network management, which at this stage can be performed remotely from an attendant station.
Table 1: IP Telephony Infrastructure Progression
|
Phase 1 |
Phase 2 |
Phase 3 |
|
Gatekeeper:
|
Gatekeeper:
|
Gatekeeper:
|
|
Directory Services:
|
Directory Services:
|
Directory Services:
|
|
Management:
|
Management:
|
Management:
|
|
|
Miscellaneous:
|
|
During phase 2, the gatekeeper can begin to undertake route optimization (not necessarily based on QoS), user name to IP/E.164 address translation, policy execution (security, fault recovery, etc.), and distributed coordination/hand-off for fault tolerance. DHCP and LDAP services can be integrated, and policy store/dissemination can be achieved through directory services. Network management would be fully integrated for performance and policy reasons, and application services can be managed. Multipoint conferencing capabilities would be introduced by means of H.323 multipoint control units (MCU). Interoperability with PBX and central office switch signaling would be provided.
During phase 3, the gatekeeper could run Java code, allowing it to receive dynamic updates to its code; it would coordinate with other gatekeepers to achieve dynamic load balancing; and it could perform QoS and policy-based routing. Directory services would be integrated on a global level. Full policy management would be implemented. Data and telephony network management would be fully integrated, and capacity planning support between the two types of networks would also be integrated.
Conclusions
It is clear from the analysis presented in this white paper that the LAN
telephony market is ripe. The required technologies are either available or soon
to become available. The large number of data networks now in operation
indicates that the infrastructure is well prepared for LAN telephony; this is
particularly timely due to recent technological advances supporting integrated
services such as broadband access technologies, QoS, and policy management
capabilities. The application examples provided here demonstrate that real
business benefits are attainable with LAN telephony technology.
Thanks to exponentially increasing use of the Internet and the World Wide Web, more and more users are becoming aware of LAN telephony. Furthermore, deregulation of the telecommunications industry has opened windows of opportunity for new competitors in the business.
As a result of all these factors, the LAN telephony market will experience explosive growth in the coming years, enabling users to achieve significant business productivity gains. Ultimately, the LAN telephony technology will become essential for core business applications. The public telephone networks will also eventually migrate into an IP-based infrastructure. With a completely integrated network, full multimedia applications will become both a reality and a basic business requirement.