Premiere Global Services (PGI) provides global business communications and technology-based solutions that enable PGI’s enterprise customers to automate, simplify and improve communications with their stakeholders. VoIP technology, which transmits voice conversations through the Internet and any other IP (Internet protocol)-based network, has been central to the conferencing part of that mission in recent years.

Since 2002, PGI has relayed conference calls from Europe and Asia Pacific back to the U.S. through VoIP technology. This has given foreign-based PGI customers highest-quality audio conferencing service with many U.S. locations at reduced costs.

A comprehensive understanding of VoIP and its capabilities reveals the advantages of deploying VoIP platforms to accommodate conferencing solutions. VoIP is the vanguard technology that assures the dependability of the on-demand communications that corporations expect at all times in a conferencing environment.

Digitally-based messaging is the lifeblood of domestic and international business today. The competitive edge goes to companies whose voice, video and data infrastructures align cleanly with the tele-networking products of their business partners to produce clear, uninterrupted and fast communications. VoIP technology is the best way to achieve that standard for voice traffic.

The generic business case for VoIP covers several key capabilities. While specific business priorities will play into any decision to implement VoIP, the benefits accruing to any VoIP installation may make it more effective than traditional analog-signal telephone networks. For instance:

• VoIP can transmit multiple calls over a single broadband-linked phone line.
  

• PSTN (public switched telephone network) options like three-way calling, call forwarding, caller ID and automatic redial are often available through VoIP, but not through local telecommunications companies (telcos).

• The VoIP architecture uses existing common protocols to establish call security.

• VoIP-based calls are connected through the Internet, giving VoIP phone users the freedom to make or receive calls anywhere.

• VoIP phones can access other Web-based services during calls, such as video, data file and message exchange—and audio conferencing.

VoIP converts analog audio signals to a digital transmission. While analog calls also involve digitizing voice communications to send them over fiber optics networks, the signal re-converts to analog once it reaches its endpoint phone. Whether or not anybody is speaking on the line, the circuit stays open in both directions. That causes the transmission to slow down considerably. But VoIP gets rid of inefficient circuit switching altogether in favor of digital packet switching, where IP packets of voice data travel through the network only when the caller is speaking. The packets don’t require a dedicated line to reach their destination, either. Instead, they go through any available open channels.

There is a code in the disassembled packet data that routes the packets to their end destination, where a computer puts the voice data back together so the listener can receive it clearly. Since VoIP can send the compressed audio over the least crowded channel, the communication is faster and more efficient than it is over analog signal networks, and many more phone calls can be handled this way.

While corporations employing a pure VoIP environment are in the minority, telcos commonly use VoIP service to establish a digital network that can be upgraded to handle multiple data-transmission formats. This flexibility to develop greater IP-based functionality is making VoIP infrastructure more attractive to businesses with audio-conferencing needs.

VoIP with SIP becoming voice communication standard

Real-Time Protocol is the standard which makes it possible for VoIP communication devices to transmit digital audio packets. Before this transmission can happen, certain call-signaling protocols have to locate the remote endpoint and determine how the voice traffic will flow between the remote and local communicating computers. H.323 and Session Initiation Protocols (SIP) have been the most popular ways to do this, with H.323 having handled the majority of VoIP deployments for voice communications over time.

H.323 and SIP both make audio and other media communication happen. While its proponents say that H.323 is designed to integrate better with legacy communication systems like PSTN, SIP and all of its non-standard variations make it easier to deploy. SIP—the VoIP solution that Premiere Global Services uses for its audio conferencing products -- is much more simplified that H.323 and was designed with IP telephony in mind. Developing new SIP-based network services, such as time-sensitive call forwarding, is a quick and easy process. Because SIP doesn’t require major hardware upgrades, new features can be installed within days instead of months. This saves time and money, since upfront and recurring service costs will be lower.

Those advantages prompted PGI to employ SIP as its VoIP protocol for audio-conferencing in 2003, and now SIP has the upper hand in the industry. New deployments routinely use SIP and many H.323 applications are converting into SIP. Today, SIP has become the accepted standard.

Only about 30% of North America’s long-distance voice-traffic currently travels over VoIP networks. In 2004, IP telephony carried only 1% of all audio conferencing minutes. The cost savings attributed to VoIP have been overstated because businesses often begin with VoIP by interfacing it to legacy phone systems that are still very viable, so there’s no immediate need to replace them with a pure VoIP infrastructure. That limits the economies VoIP consumers can derive up front. Moreover, there is no perfect VoIP application yet and VoIP faces stiff price competition from the telcos and mobile phone services.

Yet VoIP certainly is the wave of the future—and the near future at that. Momentum is building as companies continue to switch out some of their old equipment in favor of VoIP products. By 2010, it’s projected that about 50% of all long-distance voice traffic in North America will go through IP networks. VoIP-enabled audio conferencing will also grow its market share significantly by then—to 45% vs. 55% for PSTN. All major phone system manufacturers are pouring their R&D money into developing IP phone systems. As traditional PBX systems age and voice service contracts lapse, the migration over to VoIP will pick up steam. Indeed, a recent Infonetics Research study predicted that the percentage of small businesses that deploy VoIP in some way will have tripled between 2006 and 2010, while half to two-thirds of large firms will have VoIP by then.

There are several powerful reasons for the trend:

• Companies want to integrate their phone systems across different sites so that remote, traveling and in-home staff have unlimited access to the same communications network through a variety of modes, including mobile and desk phones, messaging and e-mail.

• VoIP lets company staffers tap into a suite of unified communications features—e.g., find me/follow me, directory, call routing, custom greetings, presence awareness and unified messaging. They can access all e-mail, voice and text messages from a single point—an e-mail inbox.

• There are multiple connection options for conference calling from any IP-enabled device, including a desktop IP phone, PC softphone and Instant Messaging Client. Conferees can utilize VoIP’s hybrid compatibility with PSTN to access the call from either IP or PSTN connections.

• The nature of IP telephony is that it is inherently much more reliable than PSTN. It does away with the physical phone-to-bridge connections of PSTN for packet-traversed networks that can reroute voice packets down the best path for the quickest, clearest voice delivery.

PGI has enhanced that reliability by setting up three data centers to handle voice traffic in Denver, Dallas and Atlanta. PGI allows as much traffic on its VoIP network as any two of those centers can handle, so there’s always reserve capacity to process calls if one of the centers fails. This gives VoIP a tremendous uptime advantage since failover to an alternative center happens immediately. In effect, the network becomes self-healing, with full redundancy. Even the loss of a complete site will not impact overall capacity.

Sooner rather than later, VoIP will become the dominant voice communications platform because it is the emergent technology that is removing the inherent restrictions in the traditional network systems. The features which are integral to VoIP make it much easier to establish operational efficiencies like load balancing—which distributes voice-session traffic over multiple connections to increase total available bandwidth; and disaster recovery—to ensure maximum redundancy, or back-up VoIP access, if primary equipment develops any problems. The gap in performance and capabilities between PSTN and VoIP will only get larger over time, in favor of VoIP, and there will be no advantage to leveraging outdated technology.

So the question businesses have to answer about VoIP is not whether to adopt it, but when. Some business consumers may be reluctant to re-invest in their legacy systems unless there’s a significant cost savings vs. VoIP. Yet they may be apprehensive about meshing VoIP with their existing networks because they believe the process is too complicated. As VoIP technology has matured, however, it is possible to introduce a VoIP platform that integrates seamlessly with current equipment. That, in turn, positions companies to make additional investments in new or upgraded VoIP architecture in the future—and import VoIP-enabled audio-conferencing. Beyond that, service providers like PGI have a suite of products that let VoIP-enabled companies augment voice by implementing a comprehensive, ubiquitous conferencing solution—with voice, Web and video—that is available at any time. Business VoIP phone service can grow in tandem with a company’s growth.

R&D partnership with telcos

As a pioneer in the application of VoIP to audio-conferencing, Premiere Global Services has established a technology-agnostic VoIP infrastructure which can leverage its customers’ legacy systems to deliver enterprise-class VoIP service. PGI’s VoIP carrier provides the call collection capability that allows callers to dial a PSTN number and have their voice signals enter a media gateway for conversion to an IP format. At its endpoint, the call is purely VoIP, so PGI has the versatility to accept an IP call relay from its VoIP carrier as easily as it accepts an IP handoff from an enterprise customer that uses an internal IP PBX.

While PGI works largely with one VoIP carrier, its servers and hardware are co-located in facilities that are carrier-neutral. These facilities connect the enterprise customer with PGI’s VoIP conferencing, whether or not the customer has a different standards-based VoIP network or a PSTN.

PGI has deployed VoIP this way to back haul most of its international voice call traffic to the U.S. since 2003. By the end of 2007, one billion minutes worth of calls will have passed through Premiere’s VoIP network to connect the multiple-location, no-reservation conferences of PGI’s clients. Since 2005, PGI has extended this VoIP switching-and-connection capability to its domestic conferencing calls.

The research and development responsible for the evolution of PGI’s VoiP-enabled conferencing environment is a direct result of the symbiotic working relationship the company has nurtured with its major telco partner. The collaboration gives each party a clear understanding of the other’s capabilities and limitations so that they can plot out and test next-generation VoIP architecture to meet the evolving needs of clients.

Many VoIP networks were designed for one-to-one calls, so PGI consulted with its VoIP carrier on problems like how to obtain the switching capabilities to bundle multiple calls. The two partners have de-bugged the technology so that there is voice-quality clarity without impediments like noise, echo or jitter (a condition where some voice packets are lost or delayed, creating sound gaps in the conversation and making reception very choppy).

Voice quality, reliability, scalability are top priorities

VoIP-based audio-conferencing solutions have to contain a few elements. These are:

• Top-notch audio quality. This is critical in an environment where many people are simultaneously sharing the voice transmission. PGI obtains that audio standard by using the same G711, 64-kilobit codec (the device that encodes voice signals for transmission and decodes them for listening) found in traditional PSTN communications, so the reception is just as clear. This codec expands bandwidth beyond the low-bit-rate codecs of many VoIP applications.

• Dependable performance. Corporate officials and staff expect conferencing services to be as reliably available to them as the dial tone that’s present when they pick up a desk or cell phone. For that reason, customer conferees access Premiere Global Services’ VoIP conferencing system in that same way—by dialing an 800 number and entering a pass code.

• Scalability. The PGI VoIP platform can easily scale to handle high-participation conferences of any given enterprise customer. It can deliver up to 125 conferencing ports. This capacity is necessary because there is no telling how many people will converge on a given bridge for an audio conference, particularly with on-demand conferencing, where participants can dial in at any point during the session. The number and size of conferences can vary wildly from day to day, so that historical trends are the only way to anticipate the timing and attendance at conferences.

• Participants can see as well as hear each other in a VoIP audio-conferencing environment when they join in through a PC softphone. This visual conferencing mode also gives everyone complete access to the audio host controls without having to log in to a separate application.

Ultimately, it is the quality and versatility of VoIP, not any perceived cost savings, which make the case for moving to a VoIP environment. It is true that connectivity, service, maintenance, personnel and MAC (moves, adds, changes) expenses may be less, but as the size of an organization increases, so will the upfront costs that can shrink VoIP savings. Mega-sized businesses may even find that VoIP is more expensive than what they had. Looking at VoIP as a strategic business decision and a user productivity driver makes more sense than viewing it as a money-saver. Value, not cost, is a better measure of what VoIP will be worth to a company.