The way I look at it, there are three types of video conferencing -

1. Free, such as Skype, Google Hangouts, etc.
2. “Prosumer,” or those that started as web conferencing and added video later, including WebEx, Adobe Connect, etc.
3. Enterprise-grade, like RADVISION, Cisco, Polycom, etc.

The way you and your company are using video will reflect how your organization is perceived and should very much influence which technologies you decided to implement. This decision should be based on a number of factors including:

1. How important is the quality of video?
2. How many participants are usually involved in your video interactions?
3. What other features/functions are important (e.g. content-sharing, HD communications, etc.)?
4. Do you use video to conduct important meetings/transactions?
5. Will you need to access your video interactions at a later date?

Video solutions like Skype are sufficient for casual, one-on-one communications – when you want the grandparents to see your kids, for example. But Skype is not an enterprise-grade solution. There are management issues, quality issues, limitations on the number of participants, security concerns, etc. So when you’re looking to conduct business via video, Skype is probably not your best option.

“Prosumer” solutions like Adobe Connect and WebEx are those that originally started as web conferencing tools, meaning they integrate voice (although not always over IP) with content-sharing. These solutions have recently introduced video as an add-on to their services. While presenters and others may be visible via a video stream, the human visual component is not essential in these types of solutions. They are more about active speakers and passive participants and less about interactive/collaborative communications.

Finally, there are enterprise-grade video solutions. Much like other business solutions, an investment is required. Some examples of this include wireless routers for the home that come at a fraction of the price when compared to those used in businesses. The same go for other solutions where durability, security, etc. are important or even business-critical.

But if you’re still left scratching your head looking for insight into why enterprise-grade technologies cost more than consumer and prosumer-focused solutions, here are a few more reasons:

• You have to consider the bill of materials for HD video conferencing and telepresence. The hardware used in video conferencing equipment is high quality and highly reliable. For example, the DSPs used for the encoding and decoding process, the custom hardware that is developed, the pan-tilt-zoom cameras, etc. In fact, enterprise-grade cameras are notoriously expensive – they offer features you don’t see in webcams (e.g., pan-tilt-zoom). While you don’t need this when you’re Skyping with your sister who lives out-of-state, it’s a regularly used feature in the conference room.
• Additionally, a customized solution is critical in order to ensure availability, reliability, scalability, flexibility and all the other “abilities.” Customizing a do-it-yourself video solution isn’t easy. There’s a lot involved and much to consider when setting up the system, connecting the different peripherals, finding a good echo-cancellation solution for the room you set it up in, adding remote controls, etc. AV integrators exist because it’s not an easy thing to do yourself. Personally, I can compare it to housing projects. I’m far from handy in the house – when I want something important done and can’t afford to compromise quality, I call in the experts.
• There’s the supply chain to consider as well. There are several companies between equipment manufacturer and the end customer – all of whom ultimately have to consider the bottom line. The good news is that along the way, each layer adds value in some form either through customizing a solution to the specific needs of a regional customer or by integrating applications, such as enabling integration with peripherals for telemedicine, connecting to the automation of the room system as well as providing installation services.

At the end of the day, it’s up to each decision-maker to select the video solution that best suits their professional life. However, if quality (e.g., HD audio, video and content delivery, security, etc.) matters, if you’re looking to connect with many people at one time and if visual communications is an important part of your company’s work flow and success, I guarantee the investment in enterprise-grade solutions is well worth the price.

What do you think? Have you tried consumer-based video applications in your business? I’d love to hear about your experiences.

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When employees have mobile access to corporate networks, they can transition between business and leisure seamlessly, allowing them to work from anywhere, anytime.  As work is increasingly conducted outside the office, work life and personal life begins to merge, this rapid trend toward corporate network mobility is changing the workplace, including employees’ lifestyles and business operations.

According to a recent survey by the Aberdeen Group nearly two-thirds (72%) of IT organizations reported that they support “bring your own device” initiatives. Of that group, 46% said they support any device and 26% said they only support devices from an approved company list.

The need for a BYOD Policy
Mobile devices have transformed over the years, with sophisticated features like multi-touch interfaces, dual core processors, cameras capable of recording 1080p video at 30 fps, and a front-facing VGA camera for videoconferencing. These applications run on iOS or Android operating systems, and as such, mobile devices are the new desktops.

With new and more powerful mobile devices continuously flooding the market, organisations that opt to deploy a single standard for employees will be forever behind the technology curve. At the same time, providing very employee with the latest mobile device is simply too costly and time-consuming.  Employees on the other hand, can and will either upgrade or change their mobile devices as soon as the next one is available.  As the workplace landscape across the globe changes, organisations must react to deliver the employees’ growing demands, for greater flexibility and the desired need to securely access the corporate network anywhere, anytime, on any device.

Supporting this trend and learning how to manage disparate devices across the corporate network is not something organisations should do to boost productivity; it is something they must do to stay relevant and ahead of their competitors.

The Benefits of BYOD
The integration of mobile devices into corporate network is cost-effective for an organisation, allowing them to turn mobile devices into powerful business assets whilst empowering for employees, therefore access to the corporate network must be made easy-to-use and available to all employees who need access, regardless of where they are working or on which device.

In a recent survey the most requested applications employees requested was email and document sharing followed closely by video conferencing, IM, access to company databases and seamless synchronization with other business devices.

Organisations that incorporate unified communications, like voice, presence, video conferencing, instance messaging and chat capabilities will make it easier for employees to share information, locate experts, improve decision-making and drive productivity.

In a recently published whitepaper by the Unified Communications Interoperability Forum (UCI Forum), it’s stated that “one important driver for the growth of unified communications is mobility and the remote worker. No segment is growing faster than mobile communications, and virtually every mobile device will be equipped with video chat, IM, directory, and other unified communication features within a few years.”

The Value of Mobile Unified Communications
To remain competitive organisations must find ways to allow their employees to communicate and collaborate seamlessly anytime, anywhere, and from any device.  By providing employees access to technology that supports and encourages not just communication but true collaboration, an organisation can benefit from a more flexible work environment that provides greater workforce mobility, productivity and satisfaction whilst gaining competitive advantage and staying ahead of the technology curve.

Key features required for business mobile devices include:

• Presence. This feature allows employees to see their colleagues’ status, such as availability and location, so they can readily reach them. It also supports a single number for employees across all communication devices. 
• Core business functions. Mobile devices are now integrated with email, calendars, contact lists, and directories and easily support access to relevant business applications. Access to the Internet and other data sources changes the mobile device to an essential business tool.
• Unified messaging. Integration with unified messaging is a core feature for business mobile devices and allows users to receive both voice and text messages and provides the sender to respond either by voice or email.
• Access to business applications. Companies have their own unique set of core applications required for business. Mobile devices will support easy access to these business applications, so employees have information they need available regardless of location.
• Video Integration.  Employees can enjoy exceptional high quality video, continuous presence to view multiple users simultaneously and telepresence connectivity whilst maintaining robust enterprise security, providing seamless connectivity across corporate boundaries.
  • Companies can seamlessly connect their local sales teams with product experts who can participate in sales calls and support new revenue opportunities.
  • Product groups can meet and collaborate across the globe more easily with interactive meetings that support collaborative interactive sessions from a telepresence environment to the employee mobile device.
  • Business managers, who would traditionally meet for a face-to-face meeting, can build close relationships without travel and encourages more frequent contact.
• Interoperability: Organisations have a diverse range of systems installed in their environments and in order for them to leverage existing investments, interoperability is critical to reduce complexity and operational cost.

BYOD of course creates the uncertainty and concerns by IT departments on how to manage security, device management and application management.   How can such a policy be successful with all the disparate platforms, operating systems, and user interfaces?  It can, with proper planning and mobile device management strategy (MDM).

Key consideration for a MDM strategy

1. Define requirements
2. Be selective when planning and prioritizing mobility projects.
3. Develop an acceptable user and security policy.
   • Can employees use their own mobile devices for work purposes, or will you issue company devices?
   • Which applications are required, allowed, or forbidden?
   • What procedures must they follow in case of loss or theft?
4. Adopt a device and carrier mobile management solution that lets your IT staff view, report on, restrict, and support users’ mobile capabilities.
5. Choose mobile versions of your back-end applications–or mobile solutions that integrate smoothly with existing systems–so your mobile workers can access and share data in real-time
6. To safeguard critical data, require regular backups as well as password protection and encryption on all mobile devices.
7. Install security software that can wipe a device’s memory remotely when the device connects to the corporate network after being reported lost or stolen, and/or automatically after a certain number of incorrect passwords.
8. Create and issue a mobility policy to all users
9. Constantly evaluate your mobility projects.
10. Erase and reformat mobile devices before disposing of them

Cloud-based MDM
With the increased uptake and availability of the Cloud which offers “real time” applications, such as unified communications, organisations are deploying a cloud-based solution to manage security, device management and application management for disparate mobile devices.

The advantage of a cloud-based solution allows organizations to use the infrastructure and management solution hosted in the cloud which simplifies and speeds up deployment processes without the major investment in hardware and staff investment.  The ability to deploy policy and application updates to disparate mobile devices immediately and securely will likely accelerate implementation and increase user productivity by increasing reliability, scalability and security.

A cloud-based solution should:

• Reduced risk to a great extent with a complete insight of all business operations
• Provide organisations with greater operational efficiencies with a single system to secure all mobile devices on both the company- and user-owned
• Enhance data and device security and compliance with direct visibility into mobile device-level information
• Centralized policy management across multiple platforms
• Ensure compliance to industry regulations through automated processes
• Provide a flexible and effective cost model
• Enable ongoing, seamless support for the latest devices and operating systems
• Develop and expand the cloud services as and when needed over time

The Juniper report titled, “Mobile Cloud: Smart Device Strategies for Enterprise & Consumer Markets 2011-2016,” shows spending on unified communication services – such as messaging, managed e-mail, collaboration, conferencing and IP telephony – will increase over the next five years with cloud services sales expected to reach $39bn by 2016.

Conclusion
BYOD holds tremendous promise. Whilst many organisations look at this trend as a possible way to reduce costs; the real value is in increasing employee’s satisfaction and productivity, while speeding up the rate of technology adoption into the work environment.

Organisations that lay the groundwork with comprehensive management and control of employee-owned mobile devices will be in a strong position to capitalise on productivity-enhancing services, like video collaboration, a significant new trend emerging in mobile unified communications.

The fact, there’s no stopping consumerization of mobile devices and their penetration into the work environment.  Gartner predicted that by 2014 “90% of organisations will support corporate applications on personal devices.”  Organisations must take advantage of the next wave of business growth and opportunity, and accommodate their employees as they embrace extended mobile responsibilities, or the employees will continue to find ways to do it for themselves.

Business mobility is and will become an important aspect not only on how we do business today but how business will evolve.

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IT managers used to express concern over cloud security, reliability, etc, but today, the cloud has proven itself a robust, efficient and cost-effective alternative to premises-based technologies. And while hosted solutions are adequate to meet the needs of any size organization, the cloud opens up a whole new world of opportunity for small and medium businesses that have traditionally been underserved when it comes to certain technologies – one of which is videoconferencing.

Below, we explore the top five benefits of cloud-based video conferencing for SMBs.

1.       Reduced Costs
Today’s enterprise-grade video conferencing systems are now available at a fraction of the price; there are a variety of services to choose from, including virtual rooms that can be accessed from your desktop, room-based systems that you can buy or lease, and even from your smartphone and tablet. This flexibility gives you the option to invest only in the technology you need, allowing your customers and partners to communicate with you by video at no additional cost.

2.       Reliability
Cloud-based video conferencing offers high-quality, always available video network without the overhead and complexities of managing it yourself. Built-in network intelligence and firewall traversal helps ensure you’ll have a reliable connection, regardless of network limitations.

3.       Scalability
You can now “buy by the drink instead of the bottle” without having to make large investments on video solutions. And as your business grows, your hosted video solution can grow with you. So invest in what your SMB needs now, and know that you can always increase your video usage later.

4.       Improved Productivity (and its green!)
Video conferencing has been improving the productivity of large companies for years. The ROI is proven and as we continue to make technological leaps and bounds, the demand and expectations from today’s tech-savvy populace, especially from SMBs, to be able to collaborate with workers, partners and vendors across distances in real-time is drastically increasing. By offering cloud-based video conferencing solutions, SMBs can still meet face-to-face without the need to drive to their location or step on plane – which also means that video conferencing has the added benefits of being a green technology that lowers your carbon footprint!

5.       Better Business Relationships
There’s no question that meeting face-to-face with people helps establish relationships and build rapport. Video conferencing allows you to meet “in-person” without having to travel. People who don’t use video say, “It’s so nice to finally meet you,” to those they’ve interacted with only by phone. But if you use video and you end up in the same room as your colleagues, all you here is, “It’s great to see you again!”

With video now moving to the cloud, SMBs can easily leverage cloud-based video conferencing to gain these five benefits that will ultimately increase collaboration and productivity, enhance relationships, offer demonstrated ROI and improve your business’ overall bottom line.

Have you tried out video conferencing or other technologies based in the cloud? If so, what benefits have you been experiencing? We’d love to hear your stories. 

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Using a video conference solution powered by RADVISION along with BurstPoint Networks’ video content management and streaming platform, soldiers stationed in Afghanistan were also greeted by Celtics President of Basketball Operations Danny Ainge shortly before the game. The event was both fun and heartwarming, and something our team was honored to be a part of.

As you can see from the picture here,
the JumboTron at TD Garden may well be the largest screen that remote desktop video conferencing has ever been displayed on!  The game was attended by about 18,000 fans.

A handful of our New Hampshire-based team oversaw the event. They attended in-person to make sure everything ran smoothly and as planned. This behind-the-scenes crew shared their experience with the RADVISION team. The RADVISION North America team was touched when we received their email recaps of the event. Knowing we brought such happiness to families has brought many of us to tears – what a wonderful experience it was to participate.

Frank Mehr, RADVISION vice president of technical services, Americas, supported the event.In his own words, he said, “I was proud, honored and humbled to be part of such great effort. What made the entire effort completely and totally worth it was the reaction of a young lady who saw and talked to her mom in Afghanistan after months of separation.”

The technology that brought it all together

Soldiers in Afghanistan joined the event using RADVISION SCOPIA Desktop, and those in TD Garden participated via a RADVISION SCOPIA XT1000 HD conference room system. The connection was hosted on a RADVISION SCOPIA Elite 5000 Series MCU, ensuring high quality communications despite network limitations. The event was hosted in one of RADVISION’s TrySCOPIA cloud-based virtual meeting rooms.

We were invited to host the event by our partner BurstPoint. BurstPoint’s VCP Conference Point recorded and streamed the event live to family members via the Internet or viewed during the game on the TD Garden Video Scoreboard. In addition, families who could not attend the event live view see their loved ones who attended via a private web portal.

Frank added, “This event turned out to be a great success for us, BurstPoint, the Celtics, and most importantly, for the soldiers in Kandahar and their families.”

Mike Houle, RADVISION director of technical services, worked tirelessly to bring this all together.  The New Hampshire team did a fantastic job representing RADVISION at this event. Photos are compliments of Roger Wallman, RADVISION director of product marketing.

We look forward to supporting other events like this one – it’s videoconferencing for a cause!

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The first and most important precaution that anyone can take is to turn the ‘auto-answer’ function off – especially if your systems are connected to the public internet.  IT should be aware of this, and RADVISION systems always automatically default the auto-answer function to ‘off.’ We take security very seriously, and we have several built-in security features that are seamless to end users.

But it’s also good to know how you – the end user – can help protect your video conference rooms.

Below are RADVISION’s top 5 tips for keeping your Video conferencing room secure.

1. Turn off your system when not in use – this may seem obvious, but if your system isn’t on, no one can hack their way into your board room.
2. Use a virtual room with moderator capabilities instead of point-to-point calling – using a virtual room opens up a plethora of security options that are intuitive and easy-to-use.
3.  Familiarize yourself with your virtual room options – meeting moderation functions enable you to control who accesses your virtual room:
• Anyone not invited to a meeting would not be allowed entrance into the meeting.
• You can use ad-hoc functions to lock your meeting – for example, one click of your mouse or one tap on your iPhone will lock down the meeting so no one else can enter.
• If you want to remove a participant, all you need to do is click again or swipe on our iPhone, and they are ejected from the call.
4.  Create a lobby for your meeting – no one can join without the moderator.
5. Require participants to use personal identification numbers (PINs) when entering your virtual room.

Some of these may seem intuitive, but it’s easy to forget you have these options available in your virtual meeting room. The good news is you don’t need to be an IT administrator to make sure your meeting room is secure. You just need to follow the simple steps above.

For more information on managing and securing your RADVISION virtual meeting room, click here.

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RADVISION SCOPIA Video Gateway for Microsoft Lync  is now officially qualified for deployment with Microsoft Lync 2010 and Microsoft Office Communications Server 2007 R2. In fact, the SCOPIA Video Gateway is the FIRST and ONLY device that has ever been qualified for Microsoft Lync deployments.

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Video conferencing is becoming increasingly popular, and global organizations are using the application from the boardroom to the desktop. Workers are adjusting – often happily – to this new world of collaboration as unified communications becomes a standard part of our workflow. New entrants to the workforce often expect video to be a part of their daily experience. And tablets are replacing laptops as the business travel device of choice.

At the same time, we’re seeing a tremendous increase in the number of telecommuters, a growing phenomenon of ‘bring your own  device,’ or ‘BYOD,’  in the workplace, and the overall consumerization of IT. Workers should be able to have a high quality video experience anywhere anytime. And they want this experience to be seamless, easy to use and provide the same features and benefits they experience in the office, including high definition audio, video and content-sharing and review as well as moderation capabilities.

While many experts have published video conferencing etiquette tips, few have  published recommendations for this newest addition to the world of visual communications – mobile video conferencing. Below, I’ve indicated 10 Top Tips for mobile video:

1. Learn the “ins and outs” of the solutions available today – look for those that enable multiparty calling, will connect to any standards-based video system/client, and that enable content-sharing and review.
2. Know how-to leverage multi-touch gestures, drag and drop, etc. to maximize the real estate and enhance your experience.
3. Learn layout options and check self-view so you know what those on the far end are seeing.
4. Understand that the experience varies between a smartphone and a tablet – utilize the best device for your needs.
5. Take advantage of device stands when possible to avoid “shaky camera syndrome.”
6. Mute yourself when not speaking, use a headset for optimal audio, and “mute” your camera when you’re on-the-go.
7. Ensure the room is well lit and avoid backlighting, as it will make you appear dark to those on the far end.
8. Choose a video conferencing solution that replicates human interaction best, i.e. ones that don’t keep you front and center as if you’re the presentation, when you’re not talking.
9. In-person meeting etiquette still applies. Distracting behaviors, like eating, checking email or having side conversations can take away from the true collaboration experience – so skip it.
10. Make the road work for you, and get connected! The quality of mobile video chatting today enables a much more personal experience, allowing users to improve productivity anywhere and at anytime.

I’d love to hear your feedback and what your experience has been with mobile video. And if you’re not yet connected, try out one of our virtual rooms for 30 days free of charge – just go to www.radvision.com/trymobile  to sign up.

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We are very happy to announce that SCOPIA Mobile v3 is now available on the Apple app store. SCOPIA Mobile allows anyone to connect to existing standards-based video conferencing networks from any iOS mobile device.

SCOPIA Mobile meetings deliver an enterprise-grade videoconferencing experience, including two-way audio, video, and content-sharing specifically designed to work over Wi-Fi, 3G and 4G networks. It was designed with true mobility in mind so that that you can connect from just about anywhere anytime.

Creating this type of experience for mobile devices with small screens wasn’t easy, but we are as aware as anyone of the need for such applications in today’s environment of BYOD, or bring your own device. SCOPIA Mobile provides a rich set of layouts focusing on video, content or both, allowing you to fully participate in any video meeting regardless of where life takes you.  We are very happy with the results and we’re sure you will be too.

When we developed the solution, it was critical that SCOPIA Mobile connects to and interoperates with existing enterprise video conferencing networks. The resulting application allows anyone to join the same multiparty call – including desktop apps and systems, room systems and even high-end telepresence rooms.   Additionally, SCOPIA Mobile extends to UC clients like Microsoft Lync, IBM Sametime and others using our SCOPIA gateway technology.

SCOPIA Mobile displays content originated by any standards-based device. Mobile users can easily zoom in to see every detail of the live content received. This is by far the best enterprise content delivery experience on a mobile device today.  But we didn’t stop here. We wanted to make online content consumption as intuitive as browsing music or photos. This is why we added the capability to easily go back and see cached images of content previously presented.

Content caching doesn’t require any setup, and with this feature, you can join a meeting 15 minutes late without missing a beat when it comes to data shared.

Some other useful features include the ability to:

• See the conference participant list
• Mute and un-mute noisy participants
• Invite new participants, using the enterprise directory
• Remove participants from the meeting
• Record the meeting
• And much more.

Because SCOPIA Mobile is part of the enterprise solution, it also implements role-based authentication, so only authorized users can perform these actions.

Other technical highlights

Our innovative bandwidth estimation technology, NetSense, constantly monitors the network and adjusts the connection speed accordingly before packet loss occurs.  This ensures the best quality possible, regardless of network conditions.

Mobility does not have to mean compromise. All of our mobile clients provide the best experience possible, including High Definition 720p on iPad2.

We encourage you to download the app today and start enjoying true mobile conferencing. To experience a demonstration of SCOPIA Mobile V3, please go to http://www.radvision.com/TryMobile for a free trial.

SCOPIA Mobile V3 requires SCOPIA back-end Infrastructure with Pro or Mobile licenses.  Version 7.6.1 or above of the SCOPIA Solution is needed. Customers with existing support contracts can get the latest SCOPIA Back-end firmware upgrade as part of their support agreement.

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Following my post Are OTT The New Walled Gardens? The organizers of the VoLTE 2011 conference in Paris invited me to speak about this topic in the conference and participate in a panel titled: “IMS: Why Operators Need it for Voice Service in LTE and How Can They Implement it?”

The conference spans different aspects of VoLTE, from the network to value added services. It brings together speakers and delegates from different segments including service providers such as Deutsche Telecom and vendors such as Ericsson, Alcatel-Lucent and Acme Packet. More about the sessions can be found on the agenda.

What will my session focus on?

In my session I will be talking about the coopetition of OTT service providers and “traditional” service providers. OTT is a threat to the incumbent service providers. In the eyes of the service providers, OTT are those that:

• Providing innovative services
• Offering competitive business models, mainly freemium
• Are cool [I can’t put a logical claim behind this but why would someone who practically has an unlimited voice or SMS plan use an OTT service such as Viber or WhatsApp?]

We see cases where service providers team-up with OTTs and officially launch an OTT service over their network. This happens in cases where the service provider is pushed against the wall by a competing service. A good example is the launch of FaceTime by Apple. This has caused service providers to launch an OTT video chat service such as Qik (later acquired by Skype) over their network.

Having said that, the introduction of OTT video service by the service provider is a pain reliever for their pressing problem of customer retention and need for innovative services. Yet, as for voice and SMS where the service provider will always want to maintain these core services under his control, also for video chat, as it becomes popular, the service provider will want to be in control and the one providing the service.

Questions I will address in my presentation include:

• Will the Service Providers become just Network Providers?
• What assets do they hold that will allow them to make the user chose their service vs. a free OTT service?
• What are the differentiators the service provider will be able to offer to the end user to regain their loyalty?
• What are the main drawbacks of OTT services (hint: the title of my presentation)?

We have a discount for you

Want to hear more? Interested in coming to the conference?

We managed to get a special 20% discount for the readers of our Blogs. To claim it, follow this link to RADVISION Blog readers discounted VoLTE 2011 registration page and use the password: azspevolte.

Note: We are not receiving any special benefits from the organizers based on registrations done on this promotional registration discount. We simply want our readers to come and enjoy the conference and if you are planning on coming please drop me a note so we will have a chance to meet.

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While they are doing this using open sourcing these products and providing them to the community, they are keeping a tight leash of control around the focus of these products – just ask Andreas Constantinou about how he thinks Google will be using Motorola and their Android:

“1. In the software dictatorship scenario (which we take as the default option), Google would make it untenable for OEMs not to follow its software specifications to the letter or to fork Android. Google stays true to its core ad business and divests everything apart from patents to a Taiwanese OEM wanting to break into the North America market.”

You can be sure that the second scenario will drive OEMs towards the same goal – it’s a very interesting and recommended read.

The more interesting part, is how Google are actually using these two tools to get an edge over competition. Here are a few examples.

SPDY

Google is all about speed – how fast they can serve their customers with search results and ads. That is at least how Bob Cringely sees it. In a very interesting post about how Google employs Carson’s speed, he notes how Google running hardware in less than maximum CPU speeds to gain the most out of their data centers:

“That would be Carson’s speed — the speed to get the most extra speed for the least extra cost. Or, as Carson put it, of finding “the least wasteful way of wasting.”  For aircraft the speed in question turned out to be 1.32 times the speed for most miles per gallon (the Bruguet Number). Carson’s Speed uses excess power most efficiently.

Other than three G-V’s and one Boeing 767 built for a harem, Google flies data centers, not airplanes. But Google’s situation going into its power experiment was actually very similar to aviation because it was an exercise in reducing power. Google data centers weren’t built to Bruguet specs, they were faster. Given this excess computing power that had already been paid for in capital terms, what was the most efficient way of using it? Carson’s Speed — about 43 percent power — leaving plenty of excess cycles for new services like Instant Search.”

I’d like to take it a step further: as speed is the most essential to Google, when they saw its inefficiencies, decided to do something about it. It wasn’t trying to improve HTTP by aiming for a new version of it, but rather starting from scratch and coming up with a new protocol: SPDY.

And how do you make a new protocol into a standard? With 2 years already for SPDY, it seems like it still isn’t in the IETF in any meaningful way yet. So the next best thing (or the really best thing) for Google was to make it a standard de-facto: their Chrome browser has it implemented already. And if you access any of Google’s web services, Google just might decide to use SPDY instead of HTTP to serve you (which they have started doing).

If up until now, Google used the web and speeding up the web to compete with desktop applications (mainly Microsoft), it is now able of competing directly with any other cloud based company – they are still going to be SPDYer.

And while Chrome isn’t the most used browser (yet), think about mobile now – Android is a large part of the market… where Google’s browser is used by default – and connectivity to Google’s own services on Android are controlled by Google.

WebM

When Google acquired On2 it was for their VP8 video codec. A year down the road, it renamed it as WebM and open sourced it. The next logical step was to add Chrome support to it (and get an ecosystem of companies around it).

The idea was to get rid of the patent payments on the H.264, but I think it was also to control the video parts of the web.

And again – with Chrome, Android and the ecosystem they built around it, they are in a good position to compete with others who are interested in media (think Microsoft and Apple).

WebRTC

WebRTC has a similar story to WebM. Google acquired GIPS, and then took the media engine, wrapped it up with JavaScript, called it WebRTC and pushed it as a kit to be integrated into browsers. This one provides media engine capabilities suitable for voice and video calling over the web from a browser.

As they control Chrome and Android, it can be expected to see WebRTC being integrated into them soon enough, with a clear aim of disrupting the voice and video communication market.

Without Chrome and Android, none of the initiatives above, as well as others that Google are promoting, wouldn’t come to play.

These initiatives are playing a unique role in driving Google’s innovation and competitive edge now. They give away these technologies as open source contributions, which builds an ecosystem and drives adoption. While their competitors can take these technologies just as easily – they will always be playing the game as followers in such a case.

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• Part 1: NetSense 101: Why do we need bandwidth estimation?
• Part 2: NetSense 101: Packet-loss-based Bandwidth Estimation
• Part 3: NetSense 101: Delay-based Bandwidth Estimation
• Part 4: NetSense 101: Q&A (this post)

You should know by now the two techniques of estimating bandwidth: based on packet losses and based on delays (=NetSense). This is what this series is all about.

In this post, I’d like to focus on a few questions I am usually asked about NetSense and provide the answers for them.

Is NetSense interoperable?

Yes it is. Even if only one participant in a video call uses NetSense, there will be an improvement in the experience of the call. In such a case, the person who will enjoy the increased experience will be the one running NetSense – the video he will receive will be of a higher quality.

Do I need to run H.264 or SVC to use NetSense?

No. NetSense doesn’t care about the actual media being sent – it is an algorithm that looks at incoming RTP packets and deduces out of it what the available bandwidth is. How to convey that information and change bitrates in the video encoders is up to the layers on top of NetSense.

Does NetSense work with both audio and video?

NetSense deals with finding available bandwidth from looking at incoming RTP packets. This means it has nothing to do with audio or video directly. We use it for video calls, where NetSense decides on the amount of bandwidth, and then our internal decision making processes will decide how to allocate that bandwidth between the audio, video and data channels of the calls.

I want to reduce bandwidth use. I understood that SVC is what I need. How does that fit with NetSense?

The truth is that SVC increases the bandwidth required for a point-to-point call for the same media quality. SVC is a technology that allows encoding video in layers, and it can be used for increasing the robustness of the video or to build better switching MCUs.

You might be able to use a bandwidth estimation mechanism to decide which layers out of your encoded SVC stream to send, but then – why would you encode all the layers? Better just estimate the bandwidth and tell the encoder dynamically how much bitrate it has available. This is what NetSense does.

Why do I need SVC if I have NetSense then?

The two technologies deal with two separate network problems.

• NetSense deals with congestion: packet loss caused due to too much data being sent
• SVC deals with corruption: packet loss that cannot be solved by reducing bitrate

You need both to get high video quality in all packet loss cases.

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• Part 1: NetSense 101: Why do we need bandwidth estimation?
• Part 2: NetSense 101: Packet-loss-based Bandwidth Estimation
• Part 3: NetSense 101: Delay-based Bandwidth Estimation (this post)
• Part 4: NetSense 101: Q&A

If you just bumped into this post – I suggest you read the first two parts of this series. It all boils down to this:

• We have bandwidth
• It fluctuates in its availability dynamically, each and every second
• We need to know how much of it we have to do video calls
• The most natural way of doing it is looking at packet losses, but this isn’t the best of ways
• If we could know the bandwidth we have / are going to have in a moment, it will allow us to reduce / increase our media’s bitrate and fit it to the network. This means we will have better video quality at a lower latency

How do we achieve that prescient knowledge? Being able to predict what the network is going to be like in the very near future?

We call it NetSense.

NetSense is a technique of sensing the current state of the network and estimating out of it how much effective bandwidth do we have available.

To understand how it works, think about switches and routers for a moment: Network switches have their own internal queues. They receive packets, store them internally for an instant, they decide where to route them next and then they send them out and clear them from their internal queues. If a switch gets too much packets at a given point in time – his internal queues will fill up, and he will start throwing out packets, causing a packet loss (=congestion).

As a thumb rule, the more packets a switch has in its queue, the longer these packets will take to get routed to their next hop, increasing their latency.

And here lies the whole idea of NetSense: it monitors for changes of the delay between media packets that are received, and out of it tries to deduce what happens in the switches along the route of the media. If it finds out that a switch somewhere starts accumulating packets in its internal queues – NetSense will re-estimate available bandwidth and act accordingly – without getting to the point of experiencing packet losses.

Here’s a diagram illustrating how NetSense works – you can compare it for the blueprint provided for the packet loss technique in my previous post.

What do you gain from NetSense?

• Better video experience, as there are no packet losses
• Lower latency, as NetSense tries to reduce congestion on the network

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• Part 1: NetSense 101: Why do we need bandwidth estimation?
• Part 2: NetSense 101: Packet-loss-based Bandwidth Estimation (this post)
• Part 3: NetSense 101: Delay-based Bandwidth Estimation
• Part 4: NetSense 101: Q&A

Bandwidth. We don’t have enough of it. And video calling devices need to know how much of it is available. That’s at least the gist of what I’ve written in why we need bandwidth estimation.

How do you estimate it today in most cases? Using packet loss information.

When two endpoints are engaged in a video call, they send RTP packets with the actual compressed media in them. In parallel to the RTP, there’s an additional protocol called RTCP that is used. RTCP takes care of sending control information between the endpoints that relates to the RTP media: how many packets were sent, received, lost, etc.

This information can then be used by the sender or the receiver of the media to change his behavior in real time.

In most systems, the decision is left to the receiver – he is the one that receives the data and is aware to some extent on how the link is behaving.

The receiver will be monitoring the incoming media, and if he sees that there are just too many packet losses (a heuristic that is different between vendors), it will estimate how much effective bandwidth it has and from that information ask the sender of that media to reduce the bitrate on one of the media channels. This in turn, will reduce the quality of the encoded data, reduce the frame-rate or the resolution – again, based on the sender’s own internal policies.

The diagram below illustrates the flow the receiver will use for the bandwidth estimation algorithm.

While this is the “industry standard”, there are a few soft spots with this solution that I want to point:

1. The amount of packet loss that causes re-estimation of the available bandwidth is based on a heuristic. It might not be the best one and it might not be able to distinguish between congestion and corruption types of packet losses (we want to focus on congestion with bandwidth estimation).
2. This solution is aggressive. As long as there’s no packet losses (or maybe little), we won’t reduce the bitrate. In a way, this is similar to the bufferbloat problem, where we try to send as much data as possible without taking into consideration the internal queues of all the switches and routers along the way. This in turn can increase the latency of the media.
3. We will reduce bandwidth only after a congestion already occurred and has affected the video quality. This is too late, especially when we will be requesting for I-frames in parallel (=frames which take a lot of bandwidth to encode).
4. The estimated bandwidth we will synchronize on isn’t an accurate one. trying to estimate it from packet losses means we either need to reduce bandwidth aggressively, and then we lose some of the bandwidth that is available for us by the network – or we might need to reduce it more than once to synchronize on the available bandwidth, which will lengthen the time until we get the video quality to a reasonable level yet again.

So you see: there are things we can improve. The main thing being able to “know” the available bandwidth before congestion occurs. While we haven’t yet developed our prophetic module here in RADVISION, we actually did come close when we talk about bandwidth estimation; but this will be the topic of my next post: how do we estimate bandwidth based on network delays.

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• Part 1: NetSense 101: Why do we need bandwidth estimation? (this post)
• Part 2: NetSense 101: Packet-loss-based Bandwidth Estimation
• Part 3: NetSense 101: Delay-based Bandwidth Estimation
• Part 4: NetSense 101: Q&A

We’ve written in the past about bandwidth estimation – in the post I just linked to and elsewhere. But I think it is time for a few more posts, to explain a bit more about the rationale behind our bandwidth estimation mechanism; and this time I plan on starting from the beginning – with the WHY question.

Video is a bandwidth hog. To send video in good quality you need a lot of bandwidth – usually upwards of 1 Mbps. And when video gets encoded before it is being sent, the decision is made about how much bandwidth to invest in each and every outgoing video frame. While we can always invest a lot, the question that becomes: Will the bits I am investing in my encoded video make it safely to their destination?

You might say that having a broadband connection to the internet should be enough for me not to dwell on the issue. But this is untrue – when I am working at home, there are multiple machines connected to the internet on that same connection, each doing its own tasks – passively updating my email inbox, Dropbox, Evernote and other useful cloud accounts, actively having my wife browse her Facebook account and my daughter watching YouTube songs. This means that effectively there’s less bandwidth available for my video call. And as if this is not enough – there are a lot of infrastructure (switches and routers) between me and my destination – infrastructure that caters other bandwidth hog users out there.

This boils down to a simple truth – while video calling requires both low latency and high bandwidth, both will fluctuate in their quality throughout my video call: I’ll have varying bandwidth and different data delays in that 30 minutes call I am having.

It means that my video encoder – that part that takes the raw video from the camera and compresses it before sending it to the network – needs to be flexible enough to change the way it works: increasing and reducing the amount of data it generates based on the current conditions of the network.

What will happen if my video encoder will ignore the network conditions and just continue generating a 1 Mbps bitstream for a network that has only 512 Kbps available? It will simply congest the network, causing packet losses, which in turn will kill any chances of having reasonable video quality until the network conditions improves.

My encoder needs a way to estimate the current network conditions (=estimate the available bandwidth), and from that decide how much bitrate it has and encode accordingly, reducing or increasing quality/resolution/frame rate to achieve that goal.

There are several ways of estimating that bandwidth, where the most common ones are based on packet losses. I’ll touch these techniques in my next post.

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Why does it take so much time to add new features? Why is it complex? What’s the big deal with video calling applications? Why should companies just license this technology from others and not develop it on their own?

The best answer I can provide is that video calling requires a lot of disciplines:

• You need a codec expert. Make that two – one for audio codecs and one for video codecs
• You’ll need someone who knows his voice algorithms and how to fine-tune an audio system (not necessarily the same person as the codec one)
• You’ll need someone who knows his video algorithms – one that knows the niche of real-time bidirectional video and not one that knows how to stream video over the internet (these are two distinct disciplines)
• You’ll need someone who is capable of working with DSPs – assuming you’re developing an embedded solution
• There’s this signaling stuff that needs to be handled – another discipline
• Interoperability. Different than just signaling or media or standards
• Some management stuff – SNMP, TR-069, SSH, TFTP or some other acronym that goes there

That’s a lot to chew for something that should be a feature in a product. And I haven’t touched issues of multiplatform support when you need to do tablets, phones and laptops all at once or dealing with user interfaces, hardware design and some other tasks.

While you might license the bits and pieces of the above stuff from various vendors and then integrate them – this comes at a huge cost of needing to have most of these disciplines in your enterprise to be able to do the integration properly.

Next time you think about building an application that requires video calling – may I suggest you license most of it from someone instead of developing on your own? It will let you focus on things that matter – the actual service and the user experience.

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As the need for VoLTE is growing dramatically around the world, developers are spurred to bring implementations to the market rapidly. As a result, standards are not always adhered to and chipset vendors have to endure end user experiences that leave much to be desired in terms of quality. There are heavy investments being made in developing VoLTE handsets. Unlike the global Internet, IMS/LTE networks contain additional embedded control functions geared to provide resource control, security, and QoS on the network. Providing a high level of quality to end users is of critical importance to the new VoLTE service. It is essential for service providers to maintain high quality and to ensure end user satisfaction.

When preparing new real-time voice and video over LTE, service providers must assess whether their network can provide and maintain the expected quality of service (QoS). Service provider administrators must make informed decisions regarding which network devices to deploy, the amount of bandwidth needed, and the optimal network configuration required to support voice and video over LTE technologies and to detect faults that may affect its usage. A proactive approach is needed to ensure VoLTE readiness before deployment on the network, as the success of these services depends on the ability of the network to support them.

End user-experience testing plays a critical and vital role in product development and quality assurance cycles, thereby helping equipment vendors to develop high-end VoLTE services.

What makes VoLTE testing complex and challenging? It is the fact that there are so many parameters to take into consideration, such as audio and video measurements, video telephony including motion level, codecs, IMS/LTE test case simulation with different scenarios that emulate the network the UE and the IMS core part, QoS, and interoperability test cases.

For LTE handset vendors it is important to have a feature testing system for IMS UE applications and IMS servers capable of simulating signaling and media, including end user experience and call features testing.

For IMS core vendors it is important to perform capacity testing, which typically refers to performance metrics and the simulation of thousands of handsets.

Providing a high level of quality can present a considerable challenge. It is vitally important to be able to quickly and reliably pinpoint and resolve quality-related problems as they occur. These issues are exactly at the heart of an upcoming free webinar we’re doing on “Ensuring VoLTE Conformance and User Experience” be sure to register to learn more about VoLTE testing.

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What’s RTCP? It’s a companion protocol to RTP – the one used to send and receive most media over IP these days. It gives it a lightweight control mechanism that provides feedback to the sender about the quality of the data it is sending – how much packets the remote side received, what got lost along the way – this kind of information.

Another use of RTCP is to know when a call can be dropped. In SIP, for example, call control can take place over UDP, where there is no way to know if one of the user agents crashed or dropped out of the network due to some extraneous reasons – especially when the only thing flowing between the user agents is the media. In such a case, monitoring RTCP reports can be used to determine if a call got dropped.

Today, RTCP can do much more. And by much more, I would like to shed light about two interesting IETF RFCs that deal with extensions to RTCP that make it quite an important part of real-time visual communications: RFC 3611 and RFC 4585.

RFC 3611: RTCP-XR (Extended Reports)

RTCP is a kind of a reporting protocol: it collects information and then reports it every once in a while. Problem is – it provides very little information. This often means that you can use it in a limited way to solve issues such as congestion.

To improve what you can achieve with RTCP, RTCP-XR (RFC 3611) was defined. RTCP-XR simply adds a bunch of additional information that can be collected:

• Now you can know not only how many packets were lost, but also what sequence numbers they had – which means that a sender can now try and retransmit them instead of sending a fresh new I-frame if he thinks this improves things.
• Now you can get know the jitter information on the receiver side. Or the MOS value he has for your voice (it’s a measurement of quality)
• Or a bunch of other stuff that may come useful if you want to improve the overall quality of the media.

What are the main additions of it over the basic reporting?

1. Sequence numbers of lost packets
2. Sequence numbers of duplicate packets
3. Packet receipt times
4. Receiver reference time
5. Delay since the last RTCP Receiver Report was received
6. Statistics summary
7. VoIP metrics

The exact use of each is reserved for the applications themselves, but this added information gives a lot of power for optimizing solutions within the scope of the standard in a way that maintains interoperability.

Oh – and if I haven’t mentioned it – RTCP-XR is required by IMS implementations.

RFC 4585: RTCP-FB (AVPF / Feedback)

RTP is used by SIP, H.323 and even XMPP to send the actual media over the network. Once done, all the control over the media itself – negotiation of different bitrates, request for I-frames and the likes are usually done over the signaling protocols: H.245 when it comes to H.323, INFO and SDP when it comes to SIP. While this means that mapping protocols and gatewaying between them is a hassle, there’s an even bigger implication: We want media to go point-to-point as much as possible to reduce latency and increase quality, which means we would like all control mechanisms of the media to traverse the same route – and signaling is often split from the media and takes a different route through a different set of entities over the network.

To solve these issues, RTCP-FB was defined in RFC 4585, which is also known as RTP/AVPF. This defines a way for applications to send commands over RTCP from the receiver of the media to the sender.

Most of this standard deals with how sending such messages occur within RTCP in a way that won’t break the percentage RTCP takes out of the allotted bitrate for the media – a rather complex and daunting task.

But then its main use has come out in another RFC that “sits” right on top of it: RTC 5104. While this one has a bunch of control messages that can now be sent, the most useful one out of it is known as “FIR” – Full Intra Request, which is the similar to the Video Fast Update message sent in H.323.

With RFC 4585, RTCP becomes a protocol that can be used for some intelligent communication and not just pumping media over an IP network.

Oh, and it just so happens that the IMTC’s SIP Parity AG have placed the use of this RFC as an essential part of their evolving SIP Video Profile.

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So you see, RTCP is a lot more than just an aside to RTP – it is a fully fledged protocol that is bound to be used and adopted as video calling becomes commonplace.

Make sure you take it into consideration the next time you develop a VoIP product.

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I’d like to revisit this a bit, due to some changes in the market, and explain how I see things moving forward.

Out of the many voice codecs available to us today, here are the ones that are the most interesting:

• G.722, a voice codec standardized by the ITU-T, providing an analog sound up to 7 kHz. It is a commonly used wideband codec for VoIP protocols such as H.323 and SIP.
• AAC-LD is a low delay audio codec that is defined in MPEG-4. It has the largest analog sound spectrum from all the codecs mentioned here. It is used by some of the video conferencing companies, but not all of them – some have opted not to use AAC at all while others have decided to go for AAC-LC (low complexity) instead. This makes the selection of an AAC-xx codec an issue if interoperability is what you have in mind.
• SILK is the codec used by Skype. It was unveiled by Skype a few years ago, with a promise to make it royalty free and widely available – most probably to make gatewaying from Skype to standards-based VoIP solutions easier. Fast forward to today, SILK is still mostly used by Skype.
• iSAC is GIPS own proprietary voice codec. After Google acquired GIPS and have open sourced it along with other GIPS technologies, it is freely available to others to use. While it hasn’t made significant inroads with VoIP systems, it is used by some of the messaging applications out there. Only time will tell if this will become prevalent in VoIP systems as well.
• AMR-WB, also known as G.722.2 is the voice codec of choice for wideband over cellular networks – that is, if you are developing something that has been specified and standardized for service providers – VoLTE or 3G-324M. What makes it interesting is that most mobile chipsets probably have its implementation etched into their hardware and not run as software on top – the tricky part is gaining access to it from 3^rd party applications.

To make it simple, here’s a table of these codecs:

Which one should you be using? That would depend on your setting and scenario. My priorities would be to start at G.722 for interoperability and move from there to the other options.

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In particular we discuss visual communications as a key enabler for telemedicine solutions, as after all they are all about accessibility. For this, visual communication is a great fit – it allows health organizations – hospitals, healthcare providers, experts and specialists – to supply their service to everyone, even remote patients, in a reasonable and economical manner.

For telemedicine Quality of Experience (QoE) is key. To create an effective doctor-patient scenario from a far the remote party needs to receive a clear, high definition representation of the scenario in hand. Same goes for a doctor-doctor scenarios, or any other case where physical boundaries are bridged via visual communications technology.

The ability to see the patient, as well as any instrumentation used or documents presented, without any perceived delay or network artifacts, is crucial for such a solution to be effective and trust-worthy. The same is true, of course, for the patient side, where seeing the doctor’s face is essential to build trust and confidence in cases of remote healthcare.

The importance of high definition video is clear. As details are extremely important, you have to transmit the finest quality to the other side, or the outcome of the medical procedure will be in risk. Any delay or artifact associated with the video can also jeopardize the process, and therefore must be avoided at all cost. Last but not least, you need to perfectly sync the video with the various instrumentations, to allowing the parties to collaborate and communicate with video in the background.

While QoE is traditionally associated with high cost, high end systems and costly leased lines, today’s technology allow healthcare providers to setup video infrastructures for telemedicine applications with fraction of the cost of dedicated telemedicine “rooms”, utilizing desktop and laptop computers as well as mobile devices such as mobile phones and tablets, and using the Internet as connection.

While this reduces cost and increases availability, it makes the challenge of maintaining high quality regardless of the network more significant. It also puts emphasis on the technical details of the solution: these devices must support high definition video, with a high frame rate and low delay. Fortunately, optimized video communication clients can already provide this for mobile computers and consumer devices.

With the growing popularity of desktop and mobile video calling, telemedicine is expected to become more popular and much more available, reaching anyone and everyone. This will definitely provide patients with a better quality of experience with regard to the medical treatments they are enjoying and are entitled to.

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Acme Packet’s Net-Net SMX IMS is a core session control/session border control (SBC), which enables service providers to reduce the complexity and cost of SIP multimedia services significantly. Their ecosystem comprises of various added services including video calling, for which RADVISION provides the necessary infrastructure and endpoints (video clients).

So what you get at the end of the day is a video calling service (think FaceTime) made easy: The servers (RADVISION’s SCOPIA Elite MCU and other components such as Bridgewater Systems HSS) are placed behind the Acme Packet SBC in the data center, while as the video clients – based on RADVISION’s BEEHD client framework – are running on various devices, such as laptops, mobile phones and tablets.

Of course, there’s a lot of things you can add to the experience, such as presence, instant messaging (IM), complicated call scenarios (for advanced types of services), etc., The BEEHD client frameworks support that and much more, and are fully standard and interoperable – that’s why it took no effort to integrate them into the Acme Packet ecosystem, and that’s why a customer would be totally free in choosing other components to add to the architecture, as long as they are standard-based.

Video calling is dominated today by OTT providers. This is mainly due to the fact that service providers were late to the game. However once they will join the stage, they will have the advantage of owning the network and being able to provide better control (quality of service) over the experience. This, of course, raises important questions of net neutrality and privacy, but still it would be hard for OTT to compete with the guys in charge of the pipes.

Another issue is the one I tackled here a few times, and Amir raised in his post – interoperability between services or “federation”, For video calling to become mainstream users will need a way to call using video without any regards to the service they are using. Just like with voice calls. The only ones capable of doing that are the service providers. And by federating video calling they would gain a huge advantage over the “walled gardens” of OTTs out there, which IMHO would practically kill the competition.

Bottom line – if you’re building a video calling service, being a service provider or a service developer, you should focus on the things that really matter for the long run:

• Fully standard and interoperable infrastructure and clients, supporting both SIP and H.323
• Ability to introduce new and advanced services without replacing the infrastructure/clients.
• A scalable platform for launching new IMS based services

And don’t miss the video.

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