Well, the newest IT marketing craze centers on the word “collaboration”.  The “cloud” (OK, admit it, just a server somewhere else, and not quite like turning on the tap) and the opportunities it represents to work together in spite of our whereabouts is being white boarded and shared across the globe (semi)instantaneously to board rooms and tablets alike.

Enterprise collaboration solutions (there’s that word again) are becoming available to small and medium sized businesses as both premise and cloud-based services become more affordable and simpler to implement, manage and use. The missing link is oftentimes bandwidth. Businesses may overlook the need for adequate and available bandwidth service and the associated monthly costs. In addition, as the workforce becomes more disperse with smart phones, tablets and work-at-homes, a unified communications strategy needs to be developed with both service and services in mind.

We all use collaboration tools to some extent. Good old face-to-face meetings are irreplaceable and an ad-hoc voice only conference call is a great way to get an issue resolved quickly. However, as bandwidth continues to become larger, more available and more affordable, these tried and true processes are being supplemented by video and web conferencing.

So what to do? The market changes fast. It is important to choose trusted technology partners that stay current. For example, Skype may be all you need; other times it may be woefully inadequate. The businesses that have survived the economy of the last four years have already cut where they can. They must now look to improve efficiencies and gain market share.

What used to cost a business thousands of dollars for a logo, now costs next to nothing for hundreds of logos to choose from. This year’s academy award winning documentary was funded through crowd sourcing. We baby boomers benefit from internet collaboration technologies. Can’t remember who played third base for the Baltimore Orioles in the 60’s? Just Google it on your iPad. (Brooks Robinson!) The next time you use social media for the latest news or look up something on Wikipedia, be glad that you live in an age of collaboration. It’s a great word!

When I asked who knows what the “cloud” is, I called on a kid who was the first to raise his hand. His reply was “it lets me get my pictures on my phone”. Here I was, ready to whiteboard WANs and LANs and explain data centers and wireless routers, and this kid nailed it! The client doesn’t care how it works; they just want to know what it does for them. Marketing 101!

I then talked about preparing for a sales call and how the internet can help. I explained that a salesperson’s website and social media must be professional, engaging and informative for the client to review. I also explained that a salesperson should first review the client’s website and social media for topics of common ground and for potential areas  to be of service. I asked what this might show to the client, thinking the answer would be that the client would think you made the effort to prepare. I again called on the student who raised her hand first. She answered “that you are a stalker”.

Wow, great point.  That was definitely an advanced course in salesmanship!

The moral of the story? Well for sure donate your time and get out in the community to help kids when you can. Secondly, engage the little buggers. And finally remember to know that you don’t know. I had a boss once that said as we were walking into an important sales call, “as long as they are talking, we are doing fine”. We have two ears and one mouth for a reason.  Listen and learn. Thanks, kids!

Typically, in small- to medium-sized businesses, videoconferencing systems are maintained by the staff of an outside telecommunications company. A single person is usually the key point for selling the proposal to corporate management, justifying and procuring funds, correctly applying videoconferencing technology to the site, and interfacing with end users. A problem occurs many times in properly meshing corporate management’s view of what videoconferencing should do with the day-to-day desires of end users. The deciphering of what the true corporate requirements are is the delicate task of the videoconferencing manager. For maximum productivity and best cost/performance ratio, the system functions and hardware need to be closely aligned with actual requirements.

The following parameters are the main areas where clear-cut decisions need to be made. These, of course, are to be tempered by future growth and availability of equipment and communication channels.

Decide on System Type

The first question to be answered is whether to use a limited-motion or freeze-frame system. Keep in mind that neither option will look like a Hollywood production. Limited-motion almost exclusively shows the conferees sitting on one side of a semi-circular table. Its intent is to allow intimate face-to-face meetings. In most applications the cameras are fixed and show a panoramic view of the conferees. The system operates in real time, and very little operator intervention is necessary, but it requires expensive communication channels and equipment. Freeze-frame installations typically have a remote-controlled camera for giving an initial view of the conferees. After the meeting is underway, attention is focused on graphics material placed on a special stand or viewed by the remote-control camera on a wall. An operator needs to press a button to send desired images to the other site. Freeze-frame is the most adaptable to technical meetings and is less expensive to buy and operate.

Pay Attention to Audio

Second, the type of audio system requires attention. The conversational exchange is still the most important aspect of videoconferencing. Attention to audio is often slighted until the installation is complete because it is less spectacular than video. At that point it becomes very apparent in a poorly designed system that audio is something we take for granted. Bear in mind that the audio will not have the quality of “just like” being in the room at the other site. Full-duplex audio is what we expect after using the common telephone, allowing both parties to speak and to be heard simultaneously. Because of room acoustics and the resulting feedback, this system has its limiations. Half-duplex, with its annoying cut-off, is familiar to most speakerphone users, but is usually not acceptable for videoconferencing. The best-suited audio systems are those that employ “quasi-duplex” audio. They use microphone gating, level shifting, echo suppression and interrupt capability for the best trade-off in performance.

The extent of networking is the third Question. In the past, most systems have been installed on a point A to point B basis. This has been for three reasons. The first is that videoconferencing is so new that companies want to ascertain the viability of installed systems. Second, point A to point B systems are the least complicated from both technical and user viewpoints. Last, complex network systems are much more expensive to install. Yet, as more and more companies turn to videoconferencing, increasing numbers of intra-company networks have been formed. The creation and control of a company network requires considerable effort from the telecommunications staff. The bottom line appears to be: use separate point A to point B systems unless networking is absolutely required. In any case, plan for the possibility of some kind of networking in the future. Plan Operator Control

The last major question, and an often neglected area in the planning stages of a videoconference system, is that of operator control. In actuality it is the makeup, ability and perception of the operator that dictates the requirements of the system. Because of the real-time nature of limited-motion and its use of fixed cameras, system control is reduced to just a few buttons. Thinking through all the possible algorithms of a conference alows the system planner insight into the best way to allocate functions to push buttons on a control panel. It cannot be over-emphasized how crucial this point is. In freeze-frame systems it is more acute because an operator is required to transform a static system into one that provides a continuous flow of information to keep pace with the meeting.

Every High Definition product package offered provides a full high definition tele-presence; this will allow the sharing of presentations, spreadsheets, documents, and multimedia. Included features are:

• 1280 x 720 resolution for clear details
• 30 frames per second for smooth motion
• 16 x 9 aspect ratio for a wider field of view
• H.239/Dual Stream to share PC video or multimedia
• Auxiliary audio/video/data inputs and outputs
• It’s a good time to take another look at video conferencing. LifeSize® makes communicating at a distance as natural as being in the same room. LifeSize high definition video communications deliver superior quality of experience, unique flexibility and unmatched price performance. Hold a videoconference with customers, partners, and employees around the globe – in true high definition.

LifeSize® and Pacific States Communications have joined to offer you the following high definition video conferencing solutions:

• Video Communications
• Choose your video system based on the features you want while benefiting from the HD quality experience. LifeSize offers a complete range of high definition solutions for any environment. Videoconference anytime, anywhere with LifeSize.
• Audio Communcations
• An authentic communications experience requires high definition audio. LifeSize® Phone™ is the high definition audio conference phone that delivers revolutionary audio conferencing sound quality and room coverage.
• Infrastrcuture
• LifeSize offers a variety of hardware infrastructure solutions for high definition video and audio deployments. Products include secure firewall/NAT traversal, ISDN to IP Gateway, and 12 and 24-port high definition Multipoint Control Units.
• Management
• LifeSize systems are designed to be IT friendly with plug-and-play installation and simple, intuitive operation. Our products offer intelligent management solutions for your video communications systems.
• Accessories
• LifeSize systems are designed for maximum flexibility and ease of use. LifeSize offers accessories to enhance your system and increase your deployment options.

But first, what is video conferencing? Fundamentally, this is a system that allows you to conduct meetings or training in diverse places at the same time. It incorporates audio and video, allowing you to talk and send data. This removes the barrier of distance that separates these individuals.

To make video conferencing possible, first a business needs to purchase the equipment. You also have to opt for a particular method of transmission, as the one you might be using may not be compatible with the other person.

This equipment includes: the Codec unit, the camera, microphone, video display, network connection, stands and equipment charts– if these will be portable or stationary, NT-1 and acoustic echo cancellers.

Most of these are easy to understand, so this article will just focus on the Codec, NT-1, and acoustic echo cancellers.

The Codec is considered to be the “brain and heart” of the video conference system. This is what compresses all of the data, so one can send and receive messages in real time and which can be viewed on the display device.

The NT-1 on the other hand is an electronic device otherwise called a terminal adapter that converts ISDN to your IP format. There is also the NT-3 version so instead of using a single line, one can to use three.

The acoustic echo cancellers are equipment that are typically attached to the ceiling. Basically, what they do is add both intelligibility and intensity to the resonant frequency response of your audio system.

The most important aspect of a video conferencing system which you have to understand is the network connection. There are two protocols namely the H.320 and the H.323. H.323 call signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN. A call model, similar to the ISDN call model, eases the introduction of IP telephony into existing networks of ISDN-based PBX systems, including transitions to IP-based PBXs. If the one you are talking with is using a different one, it will be impossible for you to talk with one another because while the first uses ISDN lines the latter which is much simpler uses IP lines similar to what you use to browse the web.

The good news is that you can still talk to the person on the other end using the older system by using a bridge that is able to handle these mixed protocols.

The video conferencing system of today is one of the highlights of living in the digital age. No longer do we have to send a message and wait for awhile to get a response when you can get a reply right then and there which increases productivity and saves time and resources.

The unified communications planning and implementation process is one that can change course over time to accommodate a maturing strategy. It should include an in-depth review of technologies—such as network investments and integrations to instant messaging and presence engines—to help build a cost-effective UC infrastructure.

At Pacific states Communications, we’ve consistently improved upon the UC implementation procedure and have learned a number of best practices. Initiating a UC deployment effectively typically has two primary steps:

Step No. 1: Evaluate and plan

Evaluating the technology necessary for a rollout and developing a timeline are key to the success of any UC strategy. Legacy and disparate PBX architecture was becoming unwieldy and old, and therefore difficult to support. It initially compared the risks and rewards associated with traditional PBXes versus IP PBXes and decided to move toward IP PBXes.

The second was to come up with a way to ease the enormous cultural shift resulting from the disappearance of the traditional desktop telephone. The UC strategy encompassed specific plans to address these issues, including detailed timelines for preparing the infrastructure, completing internal beta testing, preparedness and piloting activities prior to organization-wide implementation, educating employees, and completing the actual UC rollout (beginning with our corporate headquarters and largest satellite offices and finishing with our smaller offices).

Step No. 2: Upgrade the infrastructure

Every UC strategy requires certain infrastructure upgrades to enable the voice and data networks to be truly merged. Our strategy involved the purchase of a new SAN (storage area network) that could support Microsoft Exchange in addition to the OCS and voice mail environment. It could also provide more flexibility in our ability to store both voice and e-mail messages.

We upgraded our LAN (local area network) and WAN (wide area network) structures so that our IT department could further support voice and data on one network. We did this also so our employees could place voice calls across the network using SIP (Session Initiation Protocol) technology.

There are a few other items you should consider when you’re thinking about UC-related infrastructure upgrades and costs. As with any software purchase, be sure to budget for the purchase of licenses. Also, be prepared to renegotiate some of your contracts with existing voice and data service providers. If you deploy SIP and begin using SIP trunking, you potentially might no longer need PRI/BRI (Primary Rate Interface/Basic Rate Interface) connections, or two separate connections for voice and data. In addition, you can gain the ability to outsource your PSTN (public switched telephone network) connectivity to a third party.

UC is communications integrated to optimize business processes.

UC brings together the essential and suitable real-time and non-real-time communications with business processes and necessities based on presence capabilities, giving a reliable unified user interface and user experience across multiple devices and media types. Using a system of parameters, UC supports the enterprise to manage various types of communications across multiple devices and applications, while integrating with back-office applications, systems and business processes, with the goal of improving business agility and results, leading to increased revenues, decreased costs and improved customer service. As this definition suggests, UC is about solutions, not products.

But what is still unclear, especially to those who make products and to those who want to calculate the market size, is this: ”What makes up a UC solution?” Or, ”What UC components must or should be included when trying to define or count or forecast the UC market?”

A UC solution is comprised of the lowest amount of elements necessary to produce a quantifiable business process improvement based on optimizing the communication mechanism of that business process.

In that case, most analysts and vendors agree that UC is made up of a broad list of communication tools or components, including:

• • Call control and multimodal communications: this may or may not be an IP-PBX;
• • Presence: desktop, telephony, device presence, as well as a rules engine to manage access to presence information;
• • Messaging: instant messaging, email, voice mail, unified messaging, video messaging;
• • Conferencing: audio, Web and video;
• • Collaboration tools: whiteboarding, document sharing, etc.;
• • Mobility and mobile access;
• • Business process integration (sometimes called Communication Enabled Business Processes, or CEBP);
• • Telephony integration: PBX/IP PBX gateways to connect to the UC voice communications UC elements;
• • Many forms of clients and endpoints: telephones, SIP phones, softphones, wireless phones and mobile devices, soft clients (including web and/or voice portals); and
• • Speech recognition servers.

Most of the diverse components have been available for many years. The UC unified clients access one or more of these mechanisms to create the solution in a textbook, integrated approach. getting bigger, or shifting from one communication manner to another, becomes as straightforward as clicking the mouse, speaking a command, or touching a display.

All of the apparatus listed above are fundamentals of UC, but not UC in and of themselves. The driving question is: which elements must be used in order for something to be considered UC?

• • Presence capabilities, generally via a presence server
• • A UC client or unified user interface
• • Call control and/or integration with voice capabilities

While many of the other UC mechanisms, such as voice messaging or conferencing/collaboration, are important, you can have a UC solution without any of these elements. UC offerings should be flexible enough to allow companies to select the components they need, without having to pay for those that they don’t. And, this should be possible on a per-user basis in order to customize the solution for each individual user.

Some people believe that the IP PBX is a fundamental part of a UC solution. As already demonstrated by many companies, a UC solution can exist without voice functionality. Also, we believe that call control in a UC world is changing, and there are various ways in which to have voice conversations without the need for an IP PBX. More and more we’re seeing peer-to-peer communications, as well as software-based offerings that provide voice capabilities, and the role of the IP PBX will be greatly reduced in the coming years as it becomes a feature server.

Business process integration is another key element of a UC solution; communication-enabling business processes and applications generally provides a strong ROI. However, UC can provide basic user productivity benefits without necessarily integrating to specific business processes. For example, the basic “click to call” or “click to connect” capability is the most common UC application, and may or may not be tied to a business process.

To sum up, UC is more than a compilation of all of the various components – it is a way of integrating an appropriate and selective set of the various communication modes together with presence and a unified client interface, in order to optimize business processes, resulting in increased value to the organization in terms of reduced costs, increased revenues, and enhanced customer interactions and relationships.

Pacific States Communication, located on Reno Nv has over 5,000 VoIP endpoints installed with various degrees of UC implemented.

On Site VoIP

An on site VoIP system places all the equipment, including but not limited to the IP PBX, cables, routers, and other devices directly at your location. As opposed to Hosted VoIP, where the equipment is generally “cloud” based with service delivered via an internet connection.. From a cost perspective, depending on acquisition method (cash or lease) an on site system may be more expensive up front because of the capital expenditures, although on a month to month basis, it will be less costly than a hosted system.

When installing your VoIP equipment, some companies will have skilled internal staff to handle the installation, but more often the equipment will be purchased from a value added reseller or integrator, who will handle the installation process for you. This of course, will add to the overall cost.

The advantage to an on site system is primarily three fold; first you retain the maximum amount of control, second your total cost of ownership over 5 years is generally less, and third depending on hosted delivery the call quality of on-site system is consistently better..

Hosted VoIP

Hosted VoIP used to be primarily for very large companies. More recently with the greater avaialblity of cost effective large bandwidth hosted has also become viable for smaller companies. A hosted VoIP implementation is the simplest, and can often be provisioned in a single day. Because the actual equipment (except for your phones) is hosted at a third party facility, you don’t have to worry about installing and maintaining a PBX. As a result, up-front costs can be minimal as compared to purchasing the IP PBX.. Month-to-month costs may still be slightly higher than if you were to run an on-site system, but still less expensive than using the public switched telephone network (PSTN).

Because a hosted VoIP system relies on your service provider’s PBX system rather than your own in-house system, you may have access to a higher-end, powerful, and more feature-rich system handling your calls, and one that is being managed 24×7 by skilled staff. However, the quality of your calls are directly dependent on the quality of your internet service provider combined with the deployment methods of the Hosted equipment vendor. You will want to make sure you have a strong service level agreement and a guaranteed level of performance and minimal downtime, which to any business is quite valuable.

Hosted does have some advantages for small companies with multiple sites, road warriors and staff that works from home. Hosted can provide cost effective seamless call coverage with the appearance of a “large” company. Another key advantage to hosted VoIP is the ability to scale quickly in response to changing needs, or alternately, to scale downward (depending on agreement terms) if you find that you need fewer services and have less capacity requirements.

A variation of on site VoIP is a managed VoIP service, where a third party provider, which may be the VAR who did the installation or another provider, handles all of the management of your on site system, including day-to-day management such as adds/changes/moves, ongoing maintenance, and remote monitoring. Although this will add to your monthly cost, you will have the advantage of having access to skilled personnel who can handle everything on a remote basis 24 hours a day, and typically will provide you with a service level agreement.

Pacific States Communications is Northern Nevada’s leading resource for both VoIP and Hosted VoIP.

Quite a few people have used a computer and a microphone to record a their own voice or other sounds. The process involves sampling the sound that is heard by the computer at a very high rate (at least 8,000 times per second or more) and storing those “samples,” or “slices,” in memory or in a file on the computer. Each slice of audio is just a very tiny bit of the person’s voice or other sound recorded by the computer. The computer has the wherewithal to take all of those samples and play them, so that the listener can hear what was recorded.

VoIP is based on a similar idea, but the difference is that the audio samples are not stored locally on your computer. Instead, they are sent over the IP network to another computer and played there.

Of course, there is quite a bit more required in order to make VoIP work. When recording the sound slices, the computer might compress those slices so that they use less space and will concurrently record only a limited frequency range. There are a number of ways to compress audio, the algorithm for which is referred to as a “compressor/de-compressor”, or simply CODEC. Many CODECs exist for a variety of applications (e.g., movies and sound recordings) and, for VoIP, the CODECs are optimized for compressing voice, which significantly reduce the bandwidth used compared to an uncompressed audio stream. Speech CODECs are optimized to improve spoken words at the expense of sounds outside the frequency range of human speech. Recorded music and other sounds do not generally sound very good when passed through a speech CODEC, but that is perfectly acceptable for the task at hand.

Once the slice is recorded by the computer and compressed into a very small sample, they are collected together into larger chunks of data and placed into packets for transmission over the IP network. This process is referred to packetization. Generally, a single IP packet will contain 10 or more milliseconds of audio, with 20 or 30 milliseconds being most common.

Vint Cerf, who is often called the Father of the Internet, once explained packets in a way that is very easy to understand. Paraphrasing his description, he suggested that one think of a packet as a postcards sent via postal mail. A postcard contains just a limited amount of information. To deliver a very long message, one must send a lot of postcards. Of course, the post office might lose one or more postcards. One also has to assemble the received postcards in order, so some kind of mechanism must be used to properly order to postcards, such as placing a sequence number on the bottom right corner. One can think of data packets in an IP network as postcards.

Just like postcards sent via the postal system, some IP data packets get lost and the CODECs must compensate for lost packets by “filling in the gaps” with audio that is acceptable to the human ear. This process is referred to as packet-loss concealment (PLC). In some cases, packets are sent multiple times in order to overcome packet loss. This method is called, appropriately enough, redundancy. Another method to address packet loss, known as forward-error correction (FEC), is to include some information from previously transmitted packets in subsequent packets. By performing mathematical operations in a particular FEC scheme, it is possible to reconstruct a lost packet from information bits in neighboring packets.

Packets are also sometimes delayed, just as with the postcards sent through the post office. This is particularly problematic for VoIP systems, as delays in delivering a voice packet means the information is too old to play. Such old packets are simply discarded, just as if the packet was never received. This is acceptable, as the same PLC algorithms can smooth the audio to provide good audio quality.

Computers generally measure the packet delay and expect the delay to remain relatively constant, though delay can increase and decrease during the course of a conversation. Variation in delay (called jitter) is the most frustrating for IP devices. Delay, itself, just means it takes longer for the recorded voice spoken by the first person to be heard by the user on the far end. In general, good networks have an end-to-end delay of less than 100ms, though delay up to 400ms is considered acceptable (especially when using satellite systems). Jitter can result in choppy voice or temporary glitches, so VoIP devices must implement jitter buffer algorithms to compensate for jitter. Essentially, this means that a certain number of packets are queued before play-out and the queue length may be increased or decreased over time to reduce the number of discarded, late-arriving packets or to reduce “mouth to ear” delay. Such “adaptive jitter buffer” schemes are also used by CD recorders and other types of devices that deal with variable delay.

Video works in much the same way as voice. Video information received through a camera is broken into small pieces, compressed with a CODEC, placed into small packets, and transmitted over the IP network. This is one reason why VoIP is promising as a new technology: adding video or other media is relatively simple. Of course, there are certain issues that must be considered that are unique to video (e.g., frame refresh and much higher bandwidth requirements), but the basic principles of VoIP equally apply to video telephony.

Of course there is much more to VoIP than just sending the audio/video packets over the Internet. There must also be an agreed protocol for how computers find each other and how information is exchanged in order to allow packets to ultimately flow between the communicating devices. There must also be an agreed format (called payload format) for the contents of the media packets. We will describe some of the popular VoIP protocols in the next section.

Through this section, we have focused on computers that communicate with each other. However, VoIP is certainly not limited to desktop computers. VoIP is implemented in a variety of hardware devices, including IP phones, analog terminal adapters (ATAs), and gateways. In short, a large number of devices can enable VoIP communication, some of which allow one to use traditional telephone devices to interface with the IP networks: one does not have to throw out existing equipment to migrate to VoIP.