Specialist VoIP manufacturer Yealink has launched an adaptor which enables functions on its popular T26P and T28P IP telephones to be controlled via Jabra and Plantronics wireless headsets.

Approved for use by both of the market-dominating headset brands the new EHS36 offers two interfaces, enabling connection to Jabra units via an RJ45 output port and Plantronics models through a 3.5mm port.

Priced competitively it features plug and play set up to quickly and easily releases wireless functionality so that calls via the Yealink phones may be directed to, answered and terminated using the headset.

The launch is seen by Yealink as further releasing the potential of its feature rich yet affordable IP telephones, given the growing appetite for productivity enhancing wireless headsets and Jabra and Plantronics’ massive presence in the market. It is anticipating broad demand given increasing pressure to maximise efficiencies across a range of private and public sector organisations. These include government, educational, industrial and general businesses ranging from SMEs to Blue Chips and embracing specific applications such as call centres and receptions.

Further details and downloads are available from the VoIPon website on the Yealink page.

Specialist VoIP manufacturer Yealink has launched a new expansion module to extend the capabilities of its popular SIP-T26P and SIP-T28P IP phones. Yealink and Elastix signed the cooperation agreement on making Yealink the exclusive official phone brand for Elastix Certified Engineer Training (ECE Training), which indicates the strategic partnership built between Yealink and Elastix. This close cooperation guarantees the excellent training product kit and solutions for trainees.

The Yealink EXP39 module features a 160×320 graphic LCD and 20 physical keys which feature dual-colour LECs to enhance the ease with which functions are set and used.  A further 20 keys are available via a page switch feature.  In addition the ability to ‘daisy chain’ up to six modules via RJ11 line cord yields a total capacity of up to 240 programmable keys, each supporting all of the handset’s IP/PBX capabilities. These include functions such as SMS, BLF, BLA, paging, speed dialling, intercom and call transfer, park, pick up and return.

“The EXP39 makes the SIP-T26P and SIP-T28P an even more attractive choice for enterprise applications where there is a requirement to manage and monitor large call volumes,” noted Stonelu, the VP of Yealink. “Excellent features and functionality coupled with extremely competitive pricing mean this combination will really present a challenge to other manufacturers and expand opportunities for distributors in this vital market sector.”

“Elastix is a well-known IP Telephony and Unified Communications Solution. It is a great honor for Yealink to be the exclusive phone brand used in the Elastix ECE training. This cooperation is a strategic decision made between Yealink and Elastix. It will bring new opportunities for the development of both parties.” said StoneLu, the VP of Yealink. “We believe this is just the start of our cooperation, the two parties will actively expand more areas of cooperation in the future.”

“We are glad to have Yealink on board; this cooperation allows us to give more productivity to the end user experience and to all of our ECE attendants.” commented Paul Estrella, who is in charge of Elastix training courses organization.”

About Yealink

Yealink is professional designer and manufacturer of IP phones and video phones for the world-wide broadband telephony market. Yealink products are fully compatible with the SIP industry standard, and have broad interoperability with the major IP-PBX, softswitch and IMS on the market today. High-quality, easy to use and affordable price-are what Yealink strive all the time to meet.

However, he is quick to remind the group that the PBX is just one part of the system and the system must be considered as a whole.

Defining Availability

Jim first broke down the different definitions of “availability” into four categories and focused on what High Availability should mean—up time in terms of the user experience.

Normal Availability

• No special steps are taken to prevent downtime.

Redundancy

• Technology centric. It is used primarily for the IT staff as a “cover your butt” measure.

High Availability

• User centric
• Redundant servers
• Redundant environment
• Focus on processes and people to make sure proper documentation is in place, so you know what to do when downtime happens.

Disaster Recovery: When your facility is not accessible to you, how do you tell people what to you do?

If users can use the system, it’s said to be available. What is the definition of availability? What is it that needs to be available? These are important considerations in your network.

Anything less than 99.9% availability is not good. You could have 43.2 minutes of downtime per month, but no one does the math! The term “five-nines” or 99.999% would only result in 25.9 seconds of downtime per month. These are important standards to consider.

High Availability Using Sangoma’s Wanpipe

By using Sangoma’s Wanpipe® to deliver High Availability using TxTristate disruption or loss of availability can be measured in seconds.

According to Jim, configuring Wanpipe is really simple.

• In /etc/wanpipe/wanpipeX.conf change:

  FE_TXTRISTATE = NO

  to

  FE_TRXTRISTATE = YES

Note that from this point on, Wanpipe will start the card with the transmitter turned off.

• To enable the transmitter issue the following command:
  – wanpipemon -i w1g1 -c Ttxe
• To disable the transmitter you can use the following:
  – wanpipemon -i w1g1 -c Ttxd

Jim reiterates that the Sangoma card is just one part of building a high availability system. These are some of the other important considerations brainstormed by the group:

• Powered Environments—separate UPS units, electrical circuits, redundant cooling systems and generators.
• Network and Cabling—do you want to have separate switches? Don’t run every department off the same switch. Then if something happens, only half the department is gone.
• Separate Shelves—don’t stack servers. If someone spills coffee, you don’t take both servers out.
• Multiple Carriers.
• Test it regularly.

The Open Source PBX in a High Availability Environment

A redundant PRI connection used to be difficult to do in an open PBX system.

Now, it is as simple as taking the PRI Cable that’s in the system and splitting it out so that the primary server still has the point-to-point T1 connection. If there is a failure, once you have configured Linux Heartbeat, the Heartbeat system turns the transmitter on the Sangoma T1 card in the secondary server on, switches to that secondary server and starts the telephony application. Although calls in progress will be dropped, downtime will be measured in seconds. For more detailed information on how to configure the Heartbeat and Sangoma system, including handy diagrams, see a copy of Jim’s slides.

Ultimately, we learned if you want “five nines” reliability, you must be prepared to pay for it, as each level of redundancy increases the cost. Each additional “9” you require, Jim believes, doubles the cost.

Using a Sangoma card is a reliable open source solution that is one way to begin to keep costs in check, while delivering the availability you require.

This new firmware enhancement, available for downloading from Yealink’s website, includes a number of advanced features such as security protection, performance improvement and bug fixes.

Specially, the added XML-support enables customization and integration, connecting business processes and people to critical information by providing display-based access to services and applications. Users can easily access information and perform tasks. For example, use the displays on the IP Phones in hotel rooms to make dining reservations, set up wake-up calls, purchase attraction tickets, get directions, and check on flight status and so on.

Other new features:

XML Screen/Browser—XML browser is a simple sip-phone-custom browser function based on XML. With XML browser, customers can personalize their features，such as weather forecast inquiry, stock information, date inquiry, access to address book, and other functions.

Hot Desking—Hot Desking allows users to login their personal accounts on different phones anywhere. It is used in office where staffs are shifting to work especially like call center to maximize the resource.

Open VPN—Open VPN allows for remote and secure access to your network and application resources. So you can register the phone to your local office while you are on business.

There are some other significant features like 802.1x, call completion, call recording and BLF support for linekey.

The SIP-T2x was honoured with Technology Marketing Community (TMC) 2009 Innovation award and was selected as one of the Best VoIP Hardware Finalists by United Kingdom Internet Telephony Service Association (ITSPA).

Learn more about Yealink phones.

“Voice transcoding is an essential, but costly, component of most VoIP applications and networks,” said Frederic Dickey, director of product management at Sangoma. “While software solutions can reasonably handle a handful of channels on a server, our card-based solution allows for hundreds of simultaneous voice transcoding channels in a single server, without compromising on audio quality.”

Most IP telephony applications require the use of multiple types of voice codecs, which are used to digitally compress the voice signals at any given deployment. While voice signals from the Public Switched Telephony Network (PSTN) always come in the form of the G.711 codec, the VoIP terminal equipment and networks support a variety of voice codecs including G.729, G.726, AMR, etc. VoIP infrastructure most often needs to include the capability to mediate between endpoints supporting different codecs, but this ‘transcoding’ digital signal processing task is often costly, resource intensive and can affect the quality of the voice signals if it introduces too much latency or delay.

The D100 card, available in PCI and PCI Express form factors, allows to convert numerous simultaneous channels of voice communication from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. The card allows you to run 30, 90, 120, 240 or 480 channels of any-to-any voice codec conversion.

The D100 software drivers provide “plug-and-play” capabilities for both Asterisk^® and FreeSWITCH™, two leading open source telephony projects. With the compatible drivers, the open source telephony platforms can use the D100 cards as seamless voice transcoding resources. This means that existing Asterisk and FreeSWITCH applications can readily start leveraging the D100 capabilities. Further, the open source telephony software can be used as a gateway or session border controller to provide network-based transcoding services.

In open source telephony software, if a hosted PBX vendor wants to reduce the cost of bandwidth to reach customers’ SIP phones via a data connection, the cost, scalability and latency issues of software-based transcoding solutions would be prohibitive. However, the D100 card provides a reliable and cost-effective alternative.

In a contact center operation, the cost of voice transport over a data link to connect remote agents can rapidly become a sizeable part of the operational budget. Instead, the D100 card allows the contact center to use the low bit-rate codecs for connecting remote agents to centralized infrastructure and local telephony lines, thereby reducing the cost of connections to remote agents.

Starting at a list price of US$ 750, the D100 card is currently in customer trials and is expected to be generally available for purchase in the current quarter.

TXP1000 High-Density Video Transcoding Card

Octasic has designed the TXP1000 PCI Express (PCIe) plug-in cards to directly offload the host CPUs for high channel density video and voice transcoding applications. The TXP1000 cards use Octasic’s award winning Vocallo MGW (DSPs) Digital Signal Processors and media processing software stack to provide an easy to use and low power (less than 25 Watts) accelerator card in a PCI Express form factor. The company claims that the TXP1000 cards provide the highest transcoding density per watt in the industry using: H.264, MPEG4, H.263 video codecs and G711, G729, AMR voice codecs: A single TXP1000 card easily delivers 335 channels of video transcoding while offering power consumption well within the power budget of a single PCI card.

The TXP1000 cards should enable network service providers and Internet equipment manufacturers to rapidly develop and deploy efficient and high channel density video transcoding systems.

John Fry, director of business development at Octasic Inc. said in today’s press release, “Compared to standard x86 server-based solutions that suffer from serious power, cost and scalability limitations, Octasic’s TXP1000-based systems provide tangible benefits that can cut video transcoding systems capital costs in half, reduce the power consumption by a factor of 8 and still deliver 5 times more channels per unit of rack space. Octasic is unique in that it has the right combination of low power DSP technology, video expertise and communications technology heritage to successfully meet the demands of this growing market.”

TXP1000 cards can be used for any scalable media transcoding, web to mobile, IVVR (Interactive Voice & Video Response) and IP based voice and video telephony gateway systems. The use of an industry recognized form factor and an IP based host API and communications stack, means that TXP1000 cards can be easily added to existing systems to drastically increase channel density and efficiency.

Furthermore, the standard functionality also includes video and voice quality processing features such as video packet loss concealment, scaling, mixing & keying, as well as voice echo cancellation and adaptive noise reduction.

With demand for mobile video on the rise, real-time video transcoding is required to support the myriad of device types, video formats and dynamic bandwidth conditions on the open Internet and mobile networks.

Recently we had the opportunity to interview Tim Joint, Commercial Manager for Aculab.

Aculab provides IP and media processing boards, software and gateways to the global communications market. Aculab partners with VoIPon to distribute the ApplianX brand of VoIP gateways in the UK/WorldWide.

Listen to the podcast or read the transcription below.

[VoIPon]  I’m aware of Aculab but have only recently starting to hear talk of ApplianX (Pronounced: Appliance), is this a new product direction?

[Tim Joint, Commercial Manager, Aculab] Well, ApplianX is a brand started by Aculab for the enterprise market, offering telecom appliances. The products are unique in the market, because they are by in large single purpose devices, designed to address specific issues in specific environments. Not only to simplify the design of the product, but also resulting in a much more cost effective solution for both the channel partner, and the end user perspective. We think there’s much less complexity with ApplianX than there are with competing products, which seems to make both the end customers and our distributors very happy which is an important part when operating in today’s channel environment.

Specifically, ApplianX has two main products today, which are both gateways. There’s a DPNSS to Q.SIG Gateway, which is primarily for the UK market, and that enables legacy DPNSS based PBXs and services to connect to other devices or services, which don’t support DPNSS but they do support Q.SIG.

So, for example Cisco’s Call Manager that doesn’t support Q.SIG, but it does support DPNSS, which is pretty important to get to if you want to connect to PBXS over here in the UK. The second gateway takes that a little further and directly into the VoIP world by converting any of our TDM protocols into SIP. These protocols can be as we said before, DPNSS, DUS2, or perhaps the more standard Q.931 in UK, or even MFCR2 or T1 protocols for other parts of the world.

[VoIPon] Are there any disadvantages of moving into a SIP based environment?

[TJ, Aculab] Well, the early days of SIP were much like of the early days of ISDN, where interoperability issues were a main factor. But having said that, I think today, things have moved on a long way and today, certainly, with the ApplianX products, we’re confident of being able to work with most other vendor’s SIP implementations.

One area of development for SIP still would be in the area of supplementary services. For instance, such as call transfer or divert, and have direct equivalents now within SIP, although again they didn’t have in the early days.

Although sometimes again, there is more than one way to achieve those supplementary services, however in other areas those standards are still emerging.  So it’s often possible to implement a feature by using different means.  For example, our latest software release for ApplianX, actually supports one of the very low level features, so that means you can actually, when you have got a PBX connected to one of our gateways, you can use some of those lower level DPNSS features like that.

[VoIPon]   Does the move towards VoIP and UC affect ApplianX, given your obvious strengths in the older TDM-based world?

[TJ, Aculab] Well , Aculab have been involved in VoIP technology actually since the very early days they started. So we also do have considerable experience in IP as well.  Um, it is inevitable, that the voice market that we operate in moves further and further toward becoming IP centric, if you’d like.

However, given today’s financial climate, the old days of “rip and replace” as a strategy to get into new technologies, doesn’t really hold any more for a number of reasons. Users are very weary of new technologies and they have more say in what goes on with those technologies, and they have more say in those technologies being put on their PCs and telephone equipment.

Also, there are now many poor implementations of UC and VoIP. Where a migration strategy towards that would have by smoothing the transition and allowing a more controlled move into IP and that is exactly the strategy we had in mind with ApplianX. It’s basically a migration strategy that allows company to try and move into new technologies using proven products, but without having to throw away the not so inconsiderable investment in existing PBX infrastructure.

And also, the stark reality today, is that most businesses need to gain the most of their existing infrastructure, as possible.  So, customers just don’t have the resources to experiment with new technology. What they do need is a firm return on investment, and ApplianX really provides that by bridging between the IP and the traditional, old TDM world.

[VoIPon]   How do entrants into the UC market such as Microsoft and IBM affect ApplianX, for instance how do you perceive Office Communications Server?

[TJ, Aculab] Microsoft or IBM, as well as all the legacy PBX vendors, all have the potential for prospering as the UC market matures, as things progress. The key, we believe, will be in delivering those capabilities in a way that is logical, palatable, for those businesses, the end customers. I think most customers are really looking at slowly integrating the tools that come with UC today, and that’s where we come in. For instance, we’ve worked very successfully with Microsoft on many voice projects over the last ten years, including recently with Office Communication Server.

And the same arguments apply there as with all UC offerings really, is that in today’s business world, it’s unrealistic to move entirely to Communication Server without allowing for an integration path to the existing legacy PBXs that are already installed. And that’s exactly where we believe ApplianX fits today.

[VoIPon]   What about SIP trunks, can they offer cost savings and does that remove the need for gateway products?

[TJ, Aculab] Absolutely. Again, it’s almost an inevitable move for voice to become carrierless between offices and remote locations. In fact, many larger corporate companies that we’ve been talking to, such as financial institutions, etcetera, already have large data networks in place today.  And a gateway, a SIP to TDM gateway, does allow them to connect their existing PBX into IP-based data network, so there is no separate voice and data network required.

The advantage of the gateway is that it allows a gradual migration, so that the PBX and all of its associated functionality remains there for the users to benefit from.

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Voting will be open until 14th May 2010.

Vote for VoIPon in the Comms Business Awards 2010 today.

Recently, we had the chance to speak with David Rowe from the Free Telephony Project, based out of Australia, and Elektra Aichele from Freifunk.net Wireless Community Network in Germany.

Elektra and David came together about eight months ago to work on the Village Telco Project, a project to give low cost, Wi-Fi based telephony to people in the developing world.

They’ve been working as a team to develop one component of the project, called the Mesh Potato. The Mesh Potato is a Wi-Fi router with telephony that does mesh networking.

Elektra is the project’s Mesh Networking Guru and David is the telephony hardware person and actually Elektra is actually used the Mesh Potato to speak with us.

Listen to the podcast or read a summary of our call, below:

[VoIPon]: VoIPon has a great admiration and respect for open source projects. What’s particularly interesting about the Free Telephony Project is that it is not only based on open source software but the hardware is also entirely open, making the entire solution Free (as in speech). What made you start this project and what would you say was the most challenging part of it?

[David Rowe]: Well, I started the project back in 2005. The main reason was, just like a lot of projects, I had an itch I wanted to scratch. I was fascinated by the idea of doing embedded Asterisk device, a telephony product, that didn’t need a computer to make it run that would run standalone like an appliance. It had been an idea kicking around in my head for a few years, and I came across a chip the analog devices Blackfin, that was perfect for the job and I started working on the thing, evenings and nights, and eventually it turned into a full time occupation.

The idea of open hardware was, well I guess, I started as closed hardware/open software project, but I just felt like doing something a little different and um, the idea of open hardware got put into my head, through the Blackfin community. The CPU I’m using had released some open hardware designs and I thought, well let’s try to do the whole PBX system open hardware and see where it leads. I really like experimenting and playing with new ideas, and that’s just made it so much more fun and rewarding, making it open hardware.

In terms of challenging, to be honest, looking back it’s been nothing but fun. [Laughs] There’s been some hardware and software and people-ware issues, like any other project but the only thing that’d I’d say I haven’t done that I’ve found really challenging, that I really have a desire to use this project to help people in the developing world. And, um, with the Free Telephony Project hardware, the IPO4, the IPO1, etc., while they’ve been a big hit in the first world, I haven’t been able to use them to help thousands and thousands of people in the developing world and that’s a desire I still wish to fulfill, which is why I’m working on the Village Telco Project, in particular, the Mesh Potato.

[VoIPon]: Can you tell us anything more about the Free Telephony Project and what new hardware we might expect to see over the coming months?

[David Rowe]: Main my main activities have been over the last 13 months, have been developing the Mesh Potato. And we’re going to take that from Beta into a full production unit. Something that can run outdoors, it’s being made robust against environmental and human damage type conditions. So that will be the next big release.

That of course is a project I’m collaborating with other people. It’s being funded by the Shuttleworth Foundation and we’re working with a company called Atcom for the hardware manufacture. Um, in terms of the VoIP hardware coming out of the Free Telephony Project, that’s really taken on a life of its own, and I don’t have a lot to do with developing VoIP projects on a day-to-day basis. Um, but to be honest, I think that’s how it should be. I kicked off the project, I injected these free designs into the open source community and now people are running with them. There must be 20 or 25 derivative designs that have been released now from the small start we made four or five years ago. And that’s very satisfying for me.

My current interests for the free telephony project are addressing ease of use issues. We developed a great little hardware platform, that’s very low cost, but it’s still as hard to use as a full size Asterisk box. And, unless you’ve got Asterisk and Linux skills, that can be quite difficult to set up. So, I’m playing with this theme of ease of use. Can we make an Asterisk box as easy to set up as say a Wi-Fi router? So it’s software that I’m playing with at the moment, in particular, the ease of use idea.

[VoIPon]: If creating an entirely open based PBX was not enough we and a number of VoIPon customers are very impressed with the OSLEC open source high performance line echo canceller you have put together. What made you start this project and what were the main difficulties you experienced when developing OSLEC?

[David Rowe]: Well thank you. I’m very happy with the way OSLEC has turned out. The reason I started it was that I needed an echo canceller that would work with the IPO4, that’s the open hardware PBX that I played a part in developing. Asterisk, for many, many years had a big problem with software echo cancellation. And to be fair, echo cancellers are quite difficult to develop. You need some fairly specialized skills. So I decided to try an open source approach to developing an echo canceller.

As background, I have education in digital signal processing or DSP, so I understood the basics of echo cancellation. I, myself, had tried to build one several times over the past 15 years and had failed, so I realized what a challenge it would be.

So I tried something different. I set up a way, so a prototype echo canceller could be tested in live Asterisk boxes. And that I could sample all the signals flowing to and from the echo canceller and get those samples emailed back to me in the form of wave files.

Now what this let me do, was access many, many alpha and beta testers. Then take the results of their tests and feed them back into my development system here. So it was just like I had the ability to fly to Canada, and make a phone call on a Canadian phone line. Find out something that had failed, then take it back here and tested.

This was a real open source process that could never have happened in a closed source organization. The results were predictably, very fast development, very high quality code, and it’s currently in use by tens, if not hundreds of thousands of people around the world with very few problems with those that are using it.

[VoIPon]: Can you tell us a little about the Mesh Potato Project, what it is and who should be using it and how viable it is?

[Elektra Aichele]: Well the background is I actually met David in 2008 at a workshop in Capetown organized by the Shuttleworth Foundation 2008. I also met Steve Song who is directing this project. The idea is telephony for the developing world, and together with Wi-Fi, so we don’t have infrastructure, and a quick way to roll out infrastructure is to use wireless technology. And probably Mesh Technology, I bet you’re not familiar with the term.

Mesh Technology is: the Wi-Fi devices, some of them, they can operate in multi-point to multi-point peer to peer mode, so they can form a mesh and that way every node, every device acts as a relay station for all of the others in order to extend the range and the coverage of the network.

And the Mesh Potato is a wireless ATA. It’s the first wireless ATA on the market that can do Mesh Networking. So it’s intended to be placed on roofs in the developing worlds. It’s organizing itself, organizing a wireless network. And give good coverage by repeating the signal from station to station if necessary. And it features Asterisk as well.

David, you can jump in if you’d like to see some additions.

[David Rowe]: You’re doing a fine job, Elektra.

[VoIPon]: How do you guys see the market developing for these projects?

[David Rowe]: Perhaps, I’ll go first Elektra. The primary aim of the Village Telco and the Mesh Potato is to help people get low cost telephony in the developing world. So we’re looking at entering very high volume manufacture over the next 6 – 24 months.

There’s also some possibilities for the product in the first world. What we’ve got with the Mesh Potato and the Village Telco System is a way to roll out telephone networks based on Wi-Fi and very, very quickly. Um, you can switch on a Mesh Potato, tell it what its phone number is and be making phone calls within a minute.

So imagine a disaster situation like Haiti. You can deploy 1000 of these, set them up with a small battery and solar panel and you can have instant IP telephony connectivity across a disaster area. So we see quite a strong first world market as well.

Plus, there are a lot of people who really are interested in the community telephony aspect of Village Telco and Mesh Potato—the idea that you don’t need to rely on a phone company or even an ISP for your internet connectivity. I have people in my neighbourhood who’d like to use this to get around the existing phone companies if they could.

So there are quite a few different markets in both the first and the third world for the Village Telco and the Mesh Potato.

[VoIPon]: So Elektra, is there anything you’d like to add?

[Elektra Aichele]: I think we tried to do something like the OPC project did for the education of kids. Mesh Potato is a device that could help people build and deploy wireless infrastructure everywhere on the planet, and in particular areas with rough environment. We have spent a considerable amount of effort to make this, as robust as possible.

Recently, we had the opportunity to chat with Dmitry Baev, the Director of Product Development of Industry Dynamics and makers of VoiceGear—an integrated Skype gateway that is entirely plug-and-play and supports a variety of analog, digital, SIP, and Hybrid PBX systems.

Listen to the podcast or read a transcription of our call below.

[VoIPon] Please tell us why a customer would want to integrate their PBX with Skype. Is it reliable for business calls?

[Dmitry Baev, Industry Dynamics] Yes. First of all let me briefly describe what are the major use cases that we’ve seen our customers using with VoiceGear Skype Gateways. And there is about five or six primary use cases that we’ve seen. And, some of the most popular ones that we’ve seen would be something that we call Skype trunking.

By virtue of using the VoiceGear Skype Gateway with your PBX system, it allows all of your office staff to take advantage of cheap outgoing calls via Skype out. This can be calls made to other Skype users, which are absolutely free, those calls are made Skype to Skype, and also calls that are made to regular land lines and mobile numbers and those are charged at the minimal Skype out rates, which typically start at about 2 cents a minute, or there are some unlimited subscriptions that are country based.

And there are other use cases where customers are utilizing the Skype click to call button functionality. If you are some sort of a call centre, or you are an e-commerce site, then you can have a Skype click to call button placed on your website, and then any customer who comes to visit your website, clicks on the button and that makes a Skype call from the customer directly to the VoiceGear connect system, which is connected to your PBX. Then the call is transferred from Skype to your PBX and off to any department within your company.

Other use cases we’ve seen prominent with some companies is actually with their mobile workforce. So in this case, if you have staff who are travelling on business, perhaps they have Skype running on their laptops or mobile phones. And in this case, instead of having them pay for roaming calls when they want to call back to their office, or head office needs to pay long distance in order to reach out to an employee who is travelling on business, now this communication can be done Skype-to-Skype.  So a VoiceGear connect gateway which is placed in your office, can be used to in initiate a Skype call to a mobile employee who has Skype running on their laptop or mobile phone or any means that they have access to Skype.

And recently Skype has introduced services for iPhone, for Blackberry and for a number of other mobile platforms. So mobile communication is very big, and by having VoiceGear installed in your office, we are essentially creating a free link between any of your offices and the mobile workforce.

A last use case which is also very popular with larger enterprise companies is global presence.  So in this case, Skype offers virtual phone numbers around the world. And in this case, if you have offices in multiple countries, or you want to provide local dial in numbers for customers in foreign countries, you can use Skype online numbers.  So basically, any customer calling to that online number will instantly reach the VoiceGear Connect Gateway which is connected to your PBX. The call will get transferred to your PBX, as if it’s a local call, made to your local phone number, wherever your office is located.  And of course, in this case you’re saving money because you’re not paying for those incoming calls, and also it’s a local call for the customer.

[VoIPon] We understand that this product works with a company’s existing PBX using SIP. Who is the typical customer that would purchase such a system?

[DB] One thing I’d like to mention here is that we do fully support SIP with a full range of PBX systems, but it is not limited to SIP connectivity only. We in fact support analogue connections as well as ISDN PRI connections.  So in fact, VoiceGear Connect systems can work with, I would say, pretty much any PBX: SIP, analogue or digital.

And the second aspect here, who is the typical customer—this really depends on the business cases we just covered.

We’ve seen a lot of use from the university type of customers. And universities are typically using VoiceGear systems for accepting incoming Skype calls. A lot of universities want to publish their Skype account or Skype click to call buttons to let potential students get in touch with the university. Especially those where there is a large number of international students.

We’ve seen a lot of hotels and hospitality type businesses interested in the VoiceGear system.

Going beyond that, we’ve seen all types of use from various government organizations. VoIP service providers, especially those who want to give their existing subscribers access to Skype services, we’ve also seen big use from various international health care agencies.

[VoIPon] At what point does the upfront cost make sense for a business?

[DB]  So, in most cases what we’ve seen is the return on investment is typically projected to be under six months. So, even as you said, there is some up-front cost for obviously purchasing such equipment. A good thing is that the installation process is very seamless. So there is not a lot of time for installation and configuration, it’s very much straightforward.

Obviously the cost savings come from a number of places. First of all, is the free or really cheap outbound calling—integration of mobile workforce into your PBX system, that’s another big savings. In cases where organizations are deploying this type of technology to receive inbound calls, there are also big savings on 1800 numbers.

[VoIPon] What’s next for Industry Dynamics?

[DB] Yes, certainly. We’ve seen the Skype market developing over the past few years. Over the past year, we’ve seen Skype pushing heavily into the business environment, which perfectly aligns with what Industry Dynamics is doing here.

And towards that end, Skype is releasing a new service which is called Skype for SIP, and this service is oriented towards businesses. The value of this service is Skype really trying to become more of, if you will, a VoIP service provider.  Because Skype has over 500 million users, who are mostly consumer users, the next step for them would be to convert some of those consumer users who are using Skype at home, and push them to start using Skype at work.

So what Skype has done, with Skype for SIP is they’re basically allowing you to connect your VoIP enabled PBX system to Skype by virtue of having SIP trunks from the PBX to the Skype server. And this is very similar to what a VoIP service provider is.

This service is based on subscriptions so there is a cost per line every month that the customer would have to pay.

And towards that end, Industry Dynamics is actually developing a new product which is slated to come out about a month from today and this is going to be called VoiceGear Duo. It is a gateway that enables any legacy PBX system. So if you have a PBX or a key system, which has analogue ports, either analogue extensions or analogue line ports, it can be used with a Voice Gear Duo gateway to be connected to this new Skype SIP service. So this is really looking out to what is the next thing coming from Skype.

[VoIPon] We’ve been asking our podcast participants their opinion on the state of the market. What can VoIPon’s listeners expect from VoIP and unified communications in the future? Where do you think the market in general is headed?

[DB] Well, one thing for sure is that unified communication is here to stay. There is two sides to this coin. From one end, a lot of people are seeing Skype as a consumer type of service: It gives you voice, it gives you video, chat capabilities, file transfers. But on the other side of the coin, take a look and examine some of the other enterprise unified communications solutions from some of the bigger players in this industry, what you will see is that those players are really giving you all the same features.

So a really, really good high end communications system will give you voice,  perhaps it will give you video, perhaps it will have some chat capabilities where everything is built in, will give you some presence capabilities, perhaps file transfer between employees of the same company of course.

If you put these two offerings side by side, it is really interesting to see that what Skype has been doing for many many years now, is actually percolating into the larger enterprise and business environment.

So while a lot of people are still thinking of Skype as a consumer service, we really think that Skype has all the right ingredients to really become a major player in enterprise. And by working closely with Skype, Industry Dynamics is delivering the hardware which is necessary in order to be able to take advantage of the Skype services in a well managed and secure type of setting that an enterprise requires. So by virtue of having VoiceGear system installed along with your IT infrastructure, and your PBX system, you can take advantage of all the Skype features in a secure package tailored specifically for the enterprise use.

Dmitry Baev is the Director of Product Development of Industry Dynamics.