Twilio SIGNAL Conference

The Twilio SIGNAL virtual customers and developer conference has just finished and I would like to capture my impressions while they are still fresh.

I wrote about WebRTC back in 2012 and the Twilio conference was a great opportunity to check how much the industry has advanced in 8 years. I found familiar WebRTC terminology in Twilio’s Programmable Voice and Programmable Video offerings but Twilio has built a lot of functions on top of WebRTC, and now provides sophisticated APIs for developers to build voice and video applications quickly. It supports various programming languages on the application side and on the client side. Most of the implementation examples I came across were about adding voice and video capabilities to specific vertical industry applications in healthcare, finance, etc. and not about building complete collaboration applications on top of the Twilio APIs.

Many applications for the Twilio technology are not even related to voice and video but rather use Programmable Messaging (as in SMS) and Programmable Chat. The whole messaging business has been booming ever since we all started using SMS for two-factor user authentication and resetting passwords. We now can subscribe to SMS notifications for pretty much everything - tracking packages, getting updates on flight changes, getting reminders from our doctors, etc. – and Twilio is often the engine behind the message delivery. It also looks like SMS is taking the path of email and is rapidly becoming a valuable marketing channel. Since I mentioned "email", I have to admit I did not expect much innovation in email... but I was so wrong. Twilio acquired SendGrid, a company that has innovated a lot in the email area and is today behind more than 50% of the emails delivered to Inboxes around the world. SendGrid follows the Twilio API approach to communications and rounds up the Twilio offering which now includes a full set of communication channels: Email, SMS, Chat, Voice, and Video.

Twilio is also adding functionality to allow for rapid development of more complex applications such as Contact Centers. Twilio Flex builds on top of the APIs I mentioned above and allows developers to put together a contact center of a decent complexity within days or few weeks - compared to months and sometimes years for on-premise CC deployments. But the benefit of this new (programmable) approach to contact centers goes beyond the initial deployment. Business environments change fast (think COVID), and companies have to adjust their customer engagement process quickly. Regular contact centers require lengthy redesign and charge extra for supporting new configurations while Twilio has made everything very easy to program and changes of business processes can be reflected in the CC (as well as in any voice-only, video-only, chat-only environments) by editing few lines of code. Very powerful!            

Most recently, Twilio acquired IoT company Electric Imp, and used its technology to develop firmware for micro-controllers in IOT devices. It is called Microvisor and is fully managed by Twilio; this includes regular updates of the firmware and network stack to fix vulnerabilities. Developers can now build on the Twilio APIs and focus on high-value IoT functionality. I understand that the Microvisor was made possible by the brand-new Trustzone hardware function in ARM Cortex-M processor that allows communication via a lower level encrypted communication channel... back to Twilio.

To complete the SIGNAL overview, I have to add the wonderful discussions with former President Barack Obama and with the Delta Airline CEO Ed Bastian. I was prepared for the Delta Airline story: passengers clearly need a lot of SMS and email updates to navigate frequent changes and numerus updates related to travel in times of COVID. But I was not sure about the link between Twilio and Obama… It turns out that President Obama knows a lot about the software development community and even came to Silicon Valley to look for software developer talent to help getting the Affordable Care Act (ACA) web site up and running.  He talked extensively about his approach to attracting software programmers to work on important government projects such as Social Security and Veterans Administration.

In summation, the Twilio SIGNAL conference was very well organized and highly informative. My takeaway: Everything is possible if you provide stable APIs and enable software developers to build on top of them.  

Team Collaboration

As enterprises increasingly hire talent globally, building high-performance global teams has become a critical success factor. Traditional collaboration tools (email, file sharing, web conferencing, etc.) are struggling to meet the rapidly evolving requirements for seamless virtual team interaction.

A new generation of team collaboration tools has emerged since early 2014. The leading function in these new tools is the persistent conversation space (referred to as “channel” or “room”) followed by file sharing and app / bot libraries (used for example for integrations with other tools). Audio / video communication is sometimes embedded and sometimes provided via integration with another service provider.

The emerging Team Collaboration market segment is a result of the collision of two trends:

1) The rise of persistent collaboration spaces to support Agile and DevOps

The software development community uses a lot of tools (mostly in the Cloud) that make Agile software development easier: tools for tracking bugs and features (like Jira), tools for code management (like GitHub and Bitbucket), etc. Global development and operations (DevOps) teams require a persistent collaboration environment, well integrated with development tools. Slack, HipChat, and other tools are trying to meet the requirement. Since voice and video can be quite disruptive, the focus is on persistent messaging (some say “persistent chat”). Deep integration with software development tools is essential.

While the concept emerged in software development, it is transferable to other types of teams and we are already seeing options for Marketing, Creative, Project Management, etc. as well as integrations with business applications required for such teams to successfully collaborate.   

2) The rise of messaging in the UCC space

Enterprise Unified Communication shifted its focus from voice to video (video conferencing) and then to content sharing (web conferencing) in pursuit of better collaboration / higher employee productivity. The latest shift is happening right now and is about elevating messaging to a primary collaboration mode. The trend probably started first when Microsoft based Lync predecessor LCS on instant messaging and presence (rather than on email, voice, or video) but the reason for that radical shift today is the new messaging wave that is coming from the consumer space, where free messaging mobile applications led to service providers cutting the cost for SMS, and to everybody messaging everybody all the time. Employees are increasingly bringing their consumer messaging habits into the enterprise and are increasingly using messaging apps like Wechat, Whatsapp, and Facebook Messenger for business purposes…

The enterprises’ need for more control over security and account administration led to UC vendors developing messaging-focused team collaboration tools such as Cisco Spark, Microsoft Teams, Unify Circuit, and Alcatel-Lucent Rainbow. While messaging is the key functionality is the persistent collaboration spaces, UC vendors leveraged their know-how in voice and video and integrated these functions into the team collaboration tools. They also built bridges to existing (“legacy”) applications to guarantee customer investment protection and are working on customer migration stories.

Integrations with software development tools and business applications are still a challenge for the UC industry and there is considerable investment to partner / build integrations and close the gap.   

Conclusion

Enterprises are looking for ways to leverage team collaboration tools without adding complexity to their already complex collaboration environments. The market is very new and quite fragmented, with each vendor covering certain use cases a little bit better than the rest. In general, UC vendors are providing better integrations with their UC stacks while newcomers focus on developing rich integration capabilities with development tools and business applications.

The Evolution of the Collaboration Room Experience

I must admit I neglected the Video Networker blog over the past 3 years. I started a new job in 2013 and the scope was and still is much broader than video. I had many interesting topics to write about but they did not fit the Video Networker label, so I did not post. Over time I assumed that Video Network had faded away, so imagine my surprise when I checked the blog activity earlier today and realized that a lot of people still go to Video Networker for information. Most visitors come from Germany, Austria, and Saudi Arabia.

I thought about it again and expanding the scope of the blog would actually be very natural. Over the past 3 years video was absorbed in all sorts of collaboration tools while Unified Communications gradually became part of the broader Digital Workplace.  

The virtual meeting experience has improved a lot. Most meetings today require some kind of content sharing, and web conferencing has become the default way for starting a meeting. Today, I rarely get a meeting invitation without a link to a virtual meeting room.

Conferencing rooms however have not kept up. Many have only a speaker phone in the middle of the table; selected few have video conferencing systems. Bridging between video conferencing and web conferencing is still not easy. Vendors that have both types of solutions are gradually converging web conferencing rooms with video conferencing rooms but there is still work to be done on the user interface and the affordability of such solutions.  

Web conferencing vendors have tried addressing collaboration rooms by offering endpoints or rather kits to build endpoints that seamlessly connect to the web conferencing solution. But we all know that a room is very different from a screen of a desktop or a mobile device. It has multiple walls, and there are usually multiple people involved. Acoustics and view angles play a major role in a great collaboration room experience. In addition, while web conferencing and video conferencing do a good job sharing content (screens or applications), they have not been successful in enabling true collaborative work on documents, images, video clips, etc.

The industry is therefore working on alternative next-generation collaboration solutions. True collaboration in a room requires a lot of space (surface) and new ways to manipulate content. Larger monitors at affordable prices deliver the additional space (surface), and content and live video can be distributed over multiple monitors hanging on different walls, thus fully leveraging the room. Touch technology allows interacting with large screens in the same way we interact with mobile devices. So the key remaining problem is how to bring the ocean of data that we have today into the collaboration room. It is not about PowerPoint presentations and spreadsheets anymore but rather about live web pages, video feeds, and real-time analytics. Once the data is available in the room, new collaboration room technology allows for creation of new assets, white boarding, brainstorming, and for storing the results from the collaboration session, so that the next meetings can continue from where the previous meeting ended.

In the meantime, the virtual meeting camp has moved one step further and targets now continuous collaboration in and between meetings. A new generation of team collaboration tools allows for persistent collaboration via group chat and document collaboration that can start long before the first meeting and continue uninterrupted between meetings.    

And then, there are considerations about cost and investment protection. Making virtual meetings available to let’s say 100,000 enterprise users means getting 100k subscriptions from the vendor / service provider. As the technology evolves, the service changes but the cost for the enterprise remains predictable. By contrast, upgrading several thousands of conference rooms in a large enterprise is a big capital expense. There is also uncertainly about the lifespan of new technologies.

There are good reasons to think hard before making a decision on collaboration room investments, and I will continue tracking this topic.   

EC13 Part 4: Redefining The Conference Room Experience

The new conference room experience is still a work in progress, and the focus is shifting from video to content sharing and annotation.

This is the last in a four-part series covering some of the major trends I saw at Enterprise Connect Orlando. Friday I covered Virtualization, Monday was Cloud Services, and yesterday Mobility. 

Redefining the conference room experience

Video conferencing vendors have been trying to extend the video conference room experience to desktop and mobile devices for a while, and competed in this realm on video quality. Now Microsoft is taking the opposite approach: extending the desktop/mobile Lync experience to conference room. The user experience in the room is driven by the capabilities of desktop and mobile video, not the other way around.

The new approach changes the implementation priorities: ease of use is the king (the main goal is to eliminate the 5-10-minute delay of the typical video meeting due to technical issues); content sharing and whiteboarding/annotation are the most important parts of the meeting, while video quality is far lower on the requirement list. For the first time, Microsoft does not depend on third-parties for providing multipoint video: Lync 2013 supports H.264 SVC, and the Microsoft AVMCU enables multipoint video calls.

By releasing its Lync Room specification to 4 partners--Crestron, LifeSize, Polycom, and SMART--Microsoft is essentially doing a trial of its Lync Room concept. The specification is not publically available but the key requirements can be recognized in the first demos--by SMART and Crestron--at Enterprise Connect.  Addressing the trends towards smaller conference rooms, wide angle cameras are required to capture people sitting close to screen/whiteboard. Based on the notion that mechanical noises distract meeting participants, Lync Rooms use digital (not mechanical) Pan/Tilt/Zoom (PTZ) cameras. Audio elements--speakers, microphones, and stereo/mono modes--are also defined, and so is the user interface for starting a meeting. The logical split of the control functions is that meeting controls are on a small control tablet while whiteboarding/annotation actions are on the large touch screen.

From the pack, SMART came to EC best prepared with its own 109-degree camera design, multiple models--small for 6 people, medium for 12, and large for 16--and support of one or two screens. SMART leveraged its experience with whiteboarding technology to differentiate. Although the video quality was not impressive (network issues, as usual), the collaboration capabilities were superb.

Crestron opted for using an off-the-shelf Logitech camera and focused on the room control experience to differentiate its Lync Room solution--a logical approach based on Crestron's background in room control. One touch of the Crestron control unit lets users switch seamlessly between "Room Control" mode and "Lync Room Collaboration" mode.

Polycom has invested a lot in interoperability with Microsoft, including support of the Microsoft RTV video codec that enables best possible quality between older Lync clients and video endpoints. Polycom also licensed its H.264 SVC implementation to Microsoft and all other UCIF members willing to use it, a move designed to create a critical mass behind one SVC flavor. But while having the best video quality between your portfolio and Lync is a tangible competitive advantage, the concern is about the overall lower importance of video in the Lync world, where content is truly the king.

While the Lync room designs are pragmatic and will improve the conferencing room experience, the cost for building rooms is still an issue, and many small companies are looking for solutions that are a little less perfect for a lot less money. Start-up Tely Labs is trying to make video collaboration more affordable and demonstrated telyHD Business Edition, which works with Tely's own simple infrastructure supporting up to 6-way video. Their telyHD Enterprise Edition was built in partnership with BlueJeans Network, and therefore allows connecting the majority of video endpoints out there. Trying to find a market segment between free clients and $10-20K rooms, Tely has very attractive price point of $550 for the enterprise or business edition; this include camera and audio but not the cost of the video monitor.

Revolabs has developed a new generation of wideband wearable and on-table wireless microphones that improve the audio capture in conference rooms. The importance for audio in collaboration cannot be overstated, and Revolabs provides an excellent alternative to wired microphones.

Conclusion

The new conference room experience is still a work in progress, and the focus is shifting from video to content sharing and annotation. The cost of equipping conference room with video is also going down. Combined with cloud services, this will democratize video.

EC13 Report Part 3: Mobility

Mobility for UC comes in many shapes. A soft client on iOS and Android is the entry ticket, but the sky is the limit after that.

This is the third in a four-part series covering some of the major trends I saw at Enterprise Connect Orlando. Friday I covered Virtualization, and yesterday Cloud Services; tomorrow's topic will be the Conference Room Experience.

Mobility

UC clients on mobile devices have quickly become the entry ticket to mobility. Every vendor I talked to--voice, video, or other--has soft client(s). iOS and Android are the preferred OS for smart phones and tablets, followed by Windows for laptops and a mention or two of BlackBerry. However, a good mobility story rarely stops at just mobility.

Extensions to the call processing software allow switching (or shifting) sessions across any number of devices associated with a user. Alcatel-Lucent's "rapid session shift" switches between mobile phone, wired phone, soft client, or a video endpoint from partner LifeSize. There is no auto detection, and the changeover is initiated by a push of a key or tap on a screen.

Thrupoint takes session management to an entirely different level. Their Session Broker--which originates from Ubiquity--allows applications to execute during session setup, and for example, to check permissions, apply policies, and re-route sessions to another location/device. Thrupoint therefore provides a scalable platform for mobility applications.

I came across several mobility servers that allow additional automation of the call handoff process and focus on the Wi-Fi/3G/4G/GSM roaming scenario. The idea is that when the user leaves Wi-Fi coverage, the call automatically switches to VoIP over 3G/4G. If the 3G/4G service deteriorates, the call automatically switches to basic TDM GSM voice.

Aastra demonstrated the Aastra Mobility Controller (AMC) from the acquisition of Munich-based Comdasys. The mobility server is offered with the Aastra's MX-1 system (acquired from Ericsson) or as as standalone product with third-party systems.

ShoreTel has fully integrated the mobility server from the Agito acquisition, and offers it with ShoreTel IP Phone System or as a standalone product in third-party environments.

Conclusion

Mobility comes in many shapes. A soft client on iOS and Android is the entry ticket, but the sky is the limit after that. Advanced mobility functions such as session shifting and session/call handoff may become important differentiators for vendors. Reliability is still an issue, since mobility touches on several networks, and network timers do not always work exactly as expected.

EC13 Report Part 2: Cloud Services

EC13 Report: Cloud ServicesVendors are rushing into creating cloud services to reach new customers--but there are concerns to address.

Vendors are rushing into creating cloud services to reach new customers--but there are concerns to address.

This is the second in a four-part series covering some of the major trends I saw at Enterprise Connect Orlando. On Friday I covered Virtualization; tomorrow's topic will be Mobility, and Wednesday's will be the Conference Room Experience.

Cloud Services

Cloud services are popping up everywhere, reflecting the lower barrier to entry for those who want to become a service provider. There are currently several hundred Managed Service Providers (MSPs) that provide voice and video conferencing services. The market is very fragmented and the barrier to entry is relatively low, so players are trying to differentiate themselves by supporting multiple vendors, solving interoperability issues, and gaining critical mass, In addition to AVI-SPL, I met with AGT, which differentiates itself through offering software-based MCU in the cloud and offering management of video endpoints.

More and more vendors across the industry are trying to make it even easier to become an MSP--by building infrastructure for running their applications in the cloud, and by encouraging their distributors to become MSPs and resell the service. As bonus, the vendors provide management tools so that the freshly baked MSPs can measure usage and charge end users for services month by month.

The cloud allows some traditional on-premise vendors to address very small customers that cannot really afford an on-premise solution. Interactive Intelligence announced its Communications as a Service (CaaS) offering called Small Center, targeting 10-50 agents--and promising unrestricted growth up to 5,000 agents without platform change. This is adding fresh competition to Five 9, a pure-play cloud call center offering.

On the video side, Magor is repositioning itself away from telepresence and towards being a visual communication infrastructure vendor. Its new Aerus cloud service leverages sophisticated routing algorithms to enable new collaboration models, away from the traditional "endpoints calling into the bridge" model.

Since these offerings rely on Internet best effort service, some industry analysts warn against it, while others embrace the Internet quality. Unified Office is taking a different approach, focusing on engineered quality for customers (small businesses with 5-75 employees) that need more than best-effort. Unified Office installs a TCN (Total Connect Now service) box on customer premise to measure QOS and analyze IP networking problems; then works with IP networking service providers to resolve issues. Unified Office also uses an open source SBC based on Asterisk that connects to multiple SIP trunking providers and routes calls based on network quality. This reminds me of the Least Cost Routing function in PBXs; however, Unified Office does not look at cost, just network quality.

The big question is "Why do vendors roll out their own cloud services?" One school of thought is that vendors waited for service providers to develop scalable services and since this did not happen, decided to try it themselves (although it is clear that running a service is very different from making products). Another school of thought is that it is just everyone in the industry looking for room for growth by experimenting with new business models, even risking competition with SP customers.

Finally, the term "hybrid cloud" was often used in conversations at Enterprise Connect, always with different meaning. Some picture "hybrid" as a media server on premise with application server in the cloud. Other imagine some services implemented in private cloud (essentially a hosted outsourced data center) while other come from the public cloud. BroadSoft's definition, for example, is having BroadWorks system in the SP data center while getting UC services from the BroadCloud.

Conclusion

Vendors are rushing into creating cloud services to reach new customers. Small enterprise companies will benefit the most, as cloud services would allow them to use platforms they cannot afford to install on premise. One concern is the best effort quality of the Internet that most cloud services rely on. The second concern is how well vendors will perform as service providers. The third concern is about vendors competing with their service provider customers.

EC13 Report Part 1: Virtualization

EC13 Report: VirtualizationMost of the core call control vendors are virtualizing at least someof their systems; now virtualization is spreading to video elements, contact centers, SBCs and more.

Most of the core call control vendors are virtualizing at least someof their systems; now virtualization is spreading to video elements, contact centers, SBCs and more.

I have been attending Enterprise Connect and its predecessors since 1998, and my focus gradually shifted from PBXs to IP-PBXs to call centers to video conferencing and most recently to cloud services. Enterprise Connect 2013 was a unique opportunity to meet with vendors from across these communications industry segments and search for commonalities and trends.

The sheer number of vendors participating in EC requires discipline and excellent planning before the event. To get a comprehensive view of the industry developments, I met with 28 vendors in back-to-back briefings, visited the exhibits, and attended a few targeted conference sessions. The most powerful trends I discovered during the event are virtualization, cloud services, mobility, and redefining of the conference room experience. I have written a post on each of these topics, and these will run on No Jitter over the next 4 days, beginning with virtualization.

Virtualization

Virtualization--once the exclusive domain of non-real-time applications--is now becoming a cornerstone for scalable and redundant deployments in voice and video communications. Here are some of the vendors' strategies:

* Trailblazer Mitel developed a strategic relationship with VMware early on, and systematically virtualized its entire portfolio. For them, virtualization is not a feature but rather a strategy for developing new channels and reaching new customers.

* Avaya has virtualized its Avaya Communications Manager and Contact Center application as well as several other elements of the Aura architecture.

* ShoreTel has virtualized its IP Phone System and Contact Center application, but their approach to virtualization is less radical: virtual machines are positioned as a solution for larger offices, while appliances continue to be the preferred option for branch offices.

* Alcatel-Lucent OpenTouch has been virtualized and is used by partners to offer cloud services. Its release included a new licensing model and a new tool to monitor licenses.

* Siemens Enterprise Communications is in the process of certifying its OpenScape portfolio with VMware. * NEC has already virtualized UNIVERGE 3C, its Windows-based UC system from the Sphere acquisition.

So virtualization is becoming a part of most vendors' strategies for pure-IP systems. When it comes to hybrid TDM/IP, Avaya, Alcatel-Lucent, Siemens, and NEC have such hybrid systems in their portfolio but are not rushing into virtualizing them.

Virtualizing hybrid systems is not easy, and even if part of the system is virtualized, there must be a TDM/IP gateway somewhere in the network to support TDM extensions and trunks, which negates some of the virtualization benefits. Aastra MX-ONE has been virtualized, although the virtual edition is only offered on the European market for now. Aastra's mobility server (Aastra Mobility Controller = AMC) is also virtualized.

In the video conferencing world, Polycom is preparing its RMX (now RealPresence Collaboration Server) platform for deployment in virtual environments. Since server virtualization is based on x86 CPU architecture, Polycom's first step is moving from Digital Signaling Processor (DSP) to x86 CPU hardware; the result is Polycom RealPresence Collaboration Server 800s, Virtual Edition, a physical server that runs the conferencing application on x86 CPU. The resulting scalability decrease (20 transcoded HD video ports, as compared to 80 in the DSP version) is compensated by offering 60 Scalable Video Coding HD ports that require no transcoding.  While the commercial success of such a solution may be limited, it is a necessary step towards running the Polycom media processing engine on generic hypervisor. Eventually, this is the architecture required to power Polycom's Cloud Axis service.

Radvision (now part of Avaya) took a different approach with Scopia Elite 6000. While the conferencing application in the appliance runs on x86 CPU, an accelerator board handles some of the video processing. That allows the scalability to be kept up to 80 transcoded HD calls and is a small step towards the cloud future. Other solution elements such as Scopia Desktop are fully virtualized.

AVI-SPL deploys Cisco video infrastructure in virtualized environments, including applications such as VCS. It would be interesting to track if Cisco will take the Polycom or Radvision approach in getting its Codian MCUs from purpose-built DSP hardware to virtual environments.

Looking across the contact center industry to test the hypothesis that virtualization is an universal trend, I came across Five9, a cloud call center provider that uses virtualization in its network of leased data centers, and Calabrio, whose Calabrio ONE suite of contact center workforce optimization software includes call recording, quality assurance, workforce management, speech analytics, and performance-based management. For Calabrio, the only real technology challenge is virtualizing the call recording part, which requires consistent high performance and dedicated HW resources.  This was a consistent theme in all conversations around virtualization: functions/applications that require dedicated resources are hard to virtualize.

In the voice networking segment, Session Border Controller (SBC) vendors are beginning to think about virtualization, with Sonus having specific plans to virtualize its SBC products before the end of 2013. They recognize the virtualization trend but want to avoid performance limitations, especially with transcoding in SBCs.

I was not sure I would find virtualization in Adtran's portfolio but it turns out Adtran Wireless Access Controller is fully virtualized on VMware. Replacing controller hardware every time the Wi-Fi standard gets updated (remember 802.11 a, b, g, n) is not efficient and the company moved to a software-only controller that they later virtualized.

Virtualization also found its way in Extreme Networks' management application, which now runs on VMware. Extreme's policy management engine retrieves user information from the enterprise directory (Active Directory) and applies policies to each user, no matter where and how the user is connected to the network. With virtualization, Extreme has carried this concept over to virtual machines: the management application retrieves information about VMs from the VMware hypervisor and allows the administrator to create profiles per VM. As the VM moves from one server to another or from one data center to another (using the vMotion feature, for example), Extreme tracks the VM and keeps applying the same policy to it no matter the location.

Conclusion

Virtualization is not a religion, and vendors have to pick and choose which elements to virtualize in their portfolios. Media processing functions are the most challenging, since they require dedicated resources--going against the fundamentals of virtualization.

 

Controlling the Borders

After Oracle announced the acquisition of Acme Packet, a lot of people asked me about the role Acme Packet plays in the network. I started piecing together bits of information to understand the tumultuous history of session border controllers that led to the acquisition. The result is this article that hopefully explains the value of Session Border Controllers (SBCs) in the network and Acme Packet’s role in this market segment.

SBC Origin

SBCs have been around since the early days of Voice over Internet Protocol (VoIP) and have always been in the center of heated debates in the industry. The VoIP purists believed that the VoIP client should communicate directly with the VoIP proxy, so that the session can be established end-to-end, that is, between the two communication parties (clients). The model first broke when telecom-style soft switches started terminating sessions to clients and messing with the session parameters; they became so-called Back to Back User Agents (B2BUA). The benefit of B2BUAs is that they can implement telephony style features like forwarding, transfer, conference, and the “holy grail” of enterprise telephony - multiple line appearance.

Once the end-to-end session was broken however, folks decided that it can be broken several times, and put another type of session breakers – session border controllers – between VoIP clients and soft switches. VoIP purists were enraged but, as time passed, the practicality of SBCs won over the argument for clean architecture. At the end of the day, SBCs solved the most difficult problem for VoIP, firewall/NAT traversal, and the choice was really between successful call based on “bad” architecture vs. failed call following the “good” VoIP blueprint.

Over the years, I have attended a lot of IETF meetings and the issue of bad SBCs messing up the clean VoIP architecture was frequently discussed. Since IETF created SIP as a signaling standard for VoIP, the discussions there were always about SIP clients, SIP SBCs, and SIP servers. In the parallel universe of video conferencing, the ITU standard H.323 was in use long before SIP emerged. Firewall traversal was as much an issue in the H.323 world as it was in SIP; therefore, H.323 SBCs started appearing in product portfolios, mostly based on the ITU-T H.460 standard. 

SBC Functionality

The original firewall traversal function in SBCs was later enhanced with other connectivity services such as manipulation of SIP messages (for example to hide the senders address), IPv4 to IPv6 interworking (for connecting VoIP clients on IPv4 networks with VoIP clients in IPv6 networks), and SIP - H.323 translation (for example, to connect H.323 video systems with SIP video clients). Additional security services were introduced to protect the network from Denial-of-Service (DoS) and other attacks and from malformed IP packets.

More powerful SBCs could also encrypt/decrypt signaling (via TLS and IPSec) and media (SRTP). SBCs gradually evolved to enforce Quality of Service (QoS) policies (for example, to limit bandwidth or limit number of calls), perform regulatory compliance functions (emergency calls prioritization, lawful interception, etc.)

Many of the new generation of SBCs also provide built-in digital signal processors (DSPs) to provide media services, such as media transcoding. The bottom line is that SBCs started as firewall traversal tools and ended up as complex elements of the VoIP network doing pretty much everything but call control (that was prerogative of the soft switch/application server).

SBC Market Dynamic

Acme Packet was at the forefront of the SBC market, some may say they defined the market segment. In the early 2000s, service providers started offering hosted VoIP services and needed reliable and scalable SBCs for their new networks. Since tier 1 service providers usually buy from a small group of vetted vendors, Acme Packet built partnerships with Ericsson and others, and became very successful selling through such channels to service providers. Two other SBC vendors - Kagoor Networks and Jasomi Networks – were less successful. I remember meeting with these companies in the early 2000. At the time I was leading the Product Management team for Devices at Siemens, and our SIP phones always went through some kind of SBCs before connecting to SIP servers / soft switches; we therefore tested interoperability. Around 2005, the SBC market became very hot and big players started buying SBC startups. Juniper Networks acquired Kagoor in 2005. Ditech Communications acquired Jasomi in 2005 (Nuance Communications acquired Ditech in 2012). Nextone was swallowed by Nextpoint which Genband acquired in 2008. AudioCodes acquired Netrake in 2006.

Big VOIP vendors took various approaches to adding SBC functionality to their portfolios. Cisco developed its own Cisco Unified Border Element (CUBE) while Avaya acquired Sipera in 2011. In the service provider market, BroadSoft relied on Acme Packet (in fact that is how I first came across Acme Packet years ago) while others like 8x8 developed their own SBC technology.

The SBC market attracted competition. In 2010, Sonus Networks newly hired CEO Ray Dolan set the goal to make Sonus “strong #2 in the SBC space”. Sonus quickly transitioned from making soft switches to making high-end SBCs but needed medium and small sized models to complete its portfolio. In August 2012, Sonus acquired Network Equipment Technologies (NET) of Fremont, CA and – with now complete line of SBC products – is positioning itself to directly compete with Acme Packet in both the enterprise and SP markets. Sansay, the #3 in SBC market, emerged from the former Nuera Communications team in San Diego in 2002. At ITexpo 2013, Sansay won the “Best Service Provider Solution” award for their transcoding technology, integrated into the SBC product.

Through the years, I got to know two other players in the SBC market. Edgewater Networks focuses on SIP SBCs for service providers and H.323 SBCs for Polycom (Video Border Proxy = VBP portfolio), although recent version of the Polycom branded product support SIP as well. Ingate Systems focuses on SIP trunking and hosts its SIP Trunking Summit / Workshop at every IT Expo event; this event evolved into the SIP Trunking - UC Seminars. I had the pleasure to present at several Ingate events, and can say that they are very well-organized, well-attended, and very educational.

SBC’ Role in the Enterprise

While many speculate about Oracle using Acme Packet to build a stronger position in the service provider market, I see a lot of applications for SBCs in the enterprise environment.

Enterprises increasingly use services from the cloud, including cloud-based voice/video services, and need tools to control the traffic to and from the cloud. SBCs provide this session control functionality. In the new architecture, enterprise users do not access the cloud services directly but through the enterprise SBC which provides security, accounting, etc. services and, if necessary, protocol translation and media transcoding. The SBC therefore becomes the official entrance for real-time communication to the enterprise. 

In addition, SBCs are critical for enterprise VoIP deployments that require connections among existing IP-PBX systems, new deployments (such as Microsoft Lync), and other enterprise applications (such as Contact Centers). When I think about it, SBCs with H.323 support can also connect the installed base of H.323 video conferencing equipment in the enterprise. In this scenario, the SBC becomes the universal connector, the glue that connects all parts of the currently disparate enterprise communication infrastructure.   

Will WebRTC Change The Communications Industry?

Back in April 2011, I wrote about the WebRTC discussion at the 80th IETF meeting in Prague. The idea to add real-time communications (RTC), that is voice and video capabilities, to the web browser is a natural progression of Google’s “everything should be done in the browser” philosophy, so there is no surprise that Google has been driving WebRTC. One and a half years later, WebRTC has created quite a buzz in the industry and a wave of startups has developed clients and services based on WebRTC technology. I was therefore excited to meet key players at the first WebRTC Conference and Expo in San Francisco that gathered about 300 participants, and focused on two questions: a) is the WebRTC technology ready for deployment and b) what is its impact on the communications industry?

The Surprise Sponsor
No one was surprised that Google sponsored the WebRTC conference.  But pretty much everyone was surprised to see Plantronics as the second sponsor of the event. Plantronics keynote speech by Joe Burton explained why... The digital divide is getting wider – while billions of devices are connected to the Internet, many people struggle with the technology. People ask Plantronics to solve the bigger communication problem: bend the web to meet the needs of the average person. It is more than headsets; it is about wearable technology that uses sensors to measure direction, temperature, moisture, etc. parameters and adjust the communication environment. Plantronics sees WebRTC as the communication infrastructure to support wearable devices because WebRTC allows much bigger pool of web developers to build communication apps much faster and leads to democratization of communication technology.

WebRTC Vision And Realty 
WebRTC’s vision is to provide interoperable royalty-free high-quality communication between browsers. To deliver on this vision, Google acquired On2 Technologies and made the VP8 video codec available royalty-free. There are many comparisons of VP8 and H.264 video codecs and the jury is still out on which one is better but one thing is for sure: transcoding between VP8 and H.264 leads to lower video quality than any of the two codes on their own.

On the audio side, the preferred codec for WebRTC is Opus, a super-wideband codec that delivers better quality than G.711 and G.722 used in IP telephony applications today. And while there are other super-wideband codecs, including AAC-LD and G.719 used in video conferencing systems, having Opus audio in the web browser will definitely drive web developers towards using it in voice/video applications. 

Unfortunately, the codec situation has not improved since early 2011. Google still insists on royalty-free VP8 and Opus while the communication industry is still based on G. and H. codecs. Some speculate that if H.264 becomes royalty-free, the argument for VP8 would be removed, and the world could happily converge around H.264. However, with the next video codec H.265 already on the horizon, the vision of using one video codec across applications and markets sure seems unrealistic.   

WebRTC Status
WebRTC today has fairly simple functionality. First, it allows the web application to get access to the user’s microphone and video camera (using getUserMedia API in JavaScript). This is critical step since JavaScript is downloaded from the web server and the user has to explicitly allow it to interact with the camera and microphone in the user’s device. Current implementations basically ask the user for permission for every WebRTC call, and there is still no good solution that combines ease of use and privacy. The second key WebRTC function is establishing a connection between two browsers (using PeerConnection API in JavaScript). This is the equivalent of “basic call” in voice communications and is therefore pretty straightforward. The third function is data transfer (using DataChannel API in JavaScript) – this function is designed for sharing data/screen as part of a collaboration session but it has not been implemented yet. In addition, WebRTC offers noise suppression, automatic gain control, echo control, etc. functions to complete the toolbox for web developers and enable creation of fully functional web based communication clients.

Google would like to see WebRTC in all web browsers but today only Google Chrome and Mozilla Firefox have the latest WebRTC functionality. Opera browser provides partial support of WebRTC (e.g., Opera v12 supports getUserMedia). Internet Explorer does not support WebRTC; the functionality can be accessed through a Chrome frame (a Google plugin for Internet Explorer). Realizing that not all devices can run voice and video in the browser (most mobile devices simply do not have the performance to do that) Google and other vendors provide native C++ versions of WebRTC for mobile platforms. In general, developers do not want to develop separate mobile apps, so as soon as the performance is available in mobile devices, the apps will run in the browser.

Mobile
One of the biggest challenges to WebRTC is addressing mobile platforms. In general, mobile devices have lower performance than wired devices and there are concerns that running software video codec in the browser will drain the mobile device battery. Others argue that the screen and wireless radio consume so much power during a video chat that the incremental power for running a software codec is negligible. Another reason for concern is Safari's ~99% share of iOS devices. Apple supports H.264 (and not VP8) and has not yet announced any plans for WebRTC. 

In mobile networks, there are several new communication methods that are competing for attention and resources. Similar to WebRTC, Voice over LTE is a technology that enables voice and video over (mobile/4G/LTE) IP packet networks. Real-time Communication Suite (RCS) on the other hand seems to be compatible with WebRTC. RCS defines what the client should look like and what functions it should support in a SIP call. It is therefore possible to create a RCS-compliant client based on WebRTC – as demonstrated by Crocodile.

Demos
The WebRTC conference included a lot of demos: some were part of the keynotes while other ran in the expo area throughout the event. 13 demos officially competed for best demo awards. Among the winners were Vidtel (Best Conferencing), Zingaya (Ready Now Award), Acme Paket (WebRTC to IMS gateway), and Crocodile (WebRTC/RCS client). Plantronics received the Visionary Award while PubNub (a San Francisco startup) won “Best WebRTC Demo” award and was also audience favorite.

Industry Impact
The current consensus is that while WebRTC clients will become very cheap or free there will be business opportunities around infrastructure and services.

Applications for WebRTC include telephony, video conferencing, collaboration solutions with gaming, etc. The current hype is around video but most benefits are in voice apps, where WebRTC can clearly cut the cost of communication and impacts existing business models. Right now, few people see WebRTC as money making opportunity and many say that asking how to make money with WebRTC is the same as asking how to make money with Flash and JavaScript.

Several companies make money connecting video conferencing traffic from Skype and GoogleTalk to enterprise video conferencing systems and this model can be extended to new WebRTC applications. But traffic among free consumer apps (like Skype, GoogleTalk, and future WebRTC apps) does not generate a lot of revenue. There is also a concern that cheap services can completely disrupt the subscription model used for voice and video services today.

Takeaways
In many ways, the WebRTC conference reminded me of the early SIP days (10-12 years ago). Many speeches predicted that WebRTC will replace PSTN but the logic question is "Why would it be different this time?" Some cited the better starting position the industry has today - with a lot of expertise about voice and video in IETF and development community. Increasing the developer pool is indeed a strong argument for WebRTC. While moving from SS7 to SIP in the 2000’s increased the developer pool from few hundreds to few thousands (the number 6,000 SIP developers was mentioned), moving to web applications increases the developer pool to several millions (potentially 10-20 million Java developers worldwide).

On the flip side, the WebRTC functionality is still quite rudimentary – mostly basic call, not even data sharing. The demos worked quite well but interoperability is still an issue due to the lack of WebRTC support in many browsers and due to incompatibility of voice and video codecs.

For service providers, WebRTC is yet another way for third-parties to develop Over The Top (OTT) services inexpensively, and compete with alternative SP-backed approaches such as VoLTE and IMS.

Enterprises are looking for creative ideas what to do with WebRTC, and most think of contact centers. Similar to the "chat" buttons and pop-up windows that have started appearing on web pages, I expect to see "call" buttons based on WebRTC technology in 2014-15. I have some ideas about WebRTC applications behind the corporate firewall and am sure that more ideas will emerge by the next WebRTC conference in June 2013 in Atlanta, Georgia. 

To come back to the original question, the impact of the WebRTC on the communications industry will depend on finding the right applications, adding the right functionality to the toolbox, and completing the standardization activities in IETF and W3C. Only then will WebRTC gain sufficient momentum and disrupt the communications market.

New Paper “Audio Performance in Multi-Codec Telepresence Systems”

Without audio, a video call is generally useless unless the video is being used for sign language or some other special application. For many years, videoconferencing users have appreciated the superior quality of the audio call that is generally part of a video call, compared to ordinary telephony. In the past five years, several vendors have introduced multi-codec videoconferencing endpoints, widely known as telepresence systems, and these devices have taken audio to a new level for business meetings while also presenting vendors with a new set of interoperability challenges.
Andrew Davis and I published a new paper (Wainhouse Research Note) on audio performance in multi-screen (multi-codec) telepresence systems. We looked at three scenarios: point-to-point calls between systems from the same vendor, multipoint calls with equipment from the same vendor, and finally, point-to-point calls between systems from different vendors. You would be surprised how the quality of the audio changes depending on the configuration.
The paper is available to WR subscribers at http://wainhouse.com/index.php

New Paper “Video Architectures: Disruption Ahead"

Recent shifts in video network architectures promise to impact the videoconferencing world in a powerful way. While IP remains the core transport medium and H.264 remains the video algorithm of choice, interest in a switched infrastructure is coming back, but with several new twists.
Andrew Davis and I published a new paper (Wainhouse Research Note) on the impact of new video architectures on the market place. We compared traditional transcoding with layer switching and stream switching technologies.
The paper is available to WR subscribers at http://wainhouse.com/index.php

Ubiquitous Visual Communications at PTC 2012

The 34th PTC Conference (tagline "Harnessing Disruption: Global, Mobile, Social, Local") gathered about 1400 attendees from the service provider community. As expected, about half of the participants were from the USA while the other half was split among Japan, Canada, China, Australia, Singapore, etc. Over the years, I have developed deep expertise in the North American and European markets (including Eastern Europe and the Russian Federation) but travelling throughout the Asia-Pacific region has always been challenging due to its size and population distribution. PTC is therefore a great opportunity to reach service providers from the Asia-Pacific region without actually travelling to their respective countries, and the conference helps me to get global perspective on the communications market. I enjoy organizing breakout sessions at PTC and inviting high-caliber speakers to discuss hot industry topics. While my PTC 2011 session was dedicated to Unified Communications Services, this year's topic was (Video) Interoperability, Interconnection, and Sky-Rocketing Global Utility.

Telepresence Systems are not only becoming more naturally realistic, they are also becoming more inter-operable with both telepresence solutions from disparate manufacturers and other visual collaboration solutions. At the same time telepresence and video networks (enterprise and carrier overlay / converged WAN) with QoS, low-latency, and high speeds are connecting at telepresence and video exchanges which are handling IP address conflicts, security, and disparate QoS tags allowing organizations to connect with partners, vendors, and customers. This growing interoperability and inter-connection along with directories, publicly available telepresence, and improved collaborative tools is sending utility, what you can do and who you can reach, sky-rocketing! The session was prominently featured on the PTC web site and now includes the slides from the three presentations.

The session included three speakers: Damian McCabe, David Gilbert, and me. Damian McCabe is Business Head and General Manager at Bharti airtel US Global Data Business. In this role, he oversees the US business for Bharti airtel including the executive management of Wholesale, Channel, Enterprise, supplier and partner accounts. Damien talked about airtel's involvement in the Open Visual Communication Consortium (OVCC), and focused on business model and roadmap. He highlighted the importance of carrier interconnects that are being implemented right now.

The OVCC interconnects are the first time large service providers (carriers) connect their IP networks to exchange real-time traffic (video calls). It took me some time to understand the revolutionary nature of the carrier interconnects for video, since I am used to IP interconnectivity across enterprises – admittedly through firewalls, session border controllers, etc. In carrier networks, however, TDM interconnects are still used for exchanging voice calls across networks, and IP interconnects are very new.

Damian's presentation generated questions from other service providers that consider joining OVCC or are in the process of joining OVCC.

Dave Gilbert is CEO and Founder of SimpleSignal. His vision to create a disruptive communications service provider attracted a team of telecom industry veterans to develop and engineer one of the first Cloud Communications platforms designed from the ground up specifically for SMB's. Dave's presentation "Bringing Telepresence to Any Mobile Device" focused on service providers' evolution from voice to video services ("Video is the new voice"), and the increasing role of mobile devices in voice and visual communications. Dave had a live demo of high-quality video call between a soft client on an iPad tablet connected over the 3G network to an HD video system at SimpleSignal's office. Although the wireless network at PTC was overloaded, the demonstration worked very well, and the audio-video quality was excellent.

My presentation was about market trends, interoperability, and standardization. In terms of market trends, I focused on the increased use of video by information workers, increasing number of video clients, and the increased share of multi-codec systems. The increasing demand for hosted and managed services (poised to become $6.2B market in 2014, based on Wainhouse Research) creates a lot of new opportunities for service providers.

In the interoperability area, the biggest advances in 2011 were around connecting multi-codec systems, and I covered interoperability scenarios including SIP, H.323, and TIP protocols, point-to-point and multipoint configurations. I highlighted the challenges around connecting systems with different number of screens, different screen aspect ratios, and different audio capabilities (that is, different number of mono or stereo channels). I concluded my part with an update from the standardization bodies (ITU-T, IETF) and interoperability organizations (IMTC, UCIF, OVCC).

In summation, PTC 2012 was a great opportunity to meet service providers from North America and Asia-Pacific. The sessions were generous both in terms of length and in terms of breaks between sessions; that allowed for discussions to continue after the official part was completed. The idea of offering ubiquitous visual communication, that is, making video as simple and reliable as a voice call, is catching on, and increasing number of service providers have joined or are about to join OVCC to work together towards "Interoperability, Interconnection, and Sky-Rocketing Global Utility".

Cluster of Conferences around ITEXPO

Summary

ITEXPO East 2012 in Miami last week was a great opportunity to meet with customers, distributors, and vendors, get the latest updates from the industry, and communicate the latest and greatest from Polycom.
The conference part had four tracks: Communication and Collaboration (mostly discussions around UC), Customer Engagements (mostly Contact Centers and Clouds), IT2.0 (Clouds everywhere), and Next-generation Service Providers (Clouds again). So in reality ITEXPO was about UC and Clouds.

TMC CEO Rich Tehrani keeps critical mass through organizing a dozen of mini-conferences in parallel to the main ITEXPO event, in effect, creating a cluster of conferences. In addition to the more established 4GWE conference and Ingate (SIP Trunking and UC) Summit, the conference cluster now includes Cloud Communications Expo, SUITS, and several other events.

Key analysts in the industry attend ITEXPO to moderate sessions, present, meet with vendors, and get updates on their solutions.

Finally, there are lively exhibits with a lot of new companies showing products and services. Traditional enterprise communication vendors (Avaya, Cisco, Siemens …) have withdrawn from the exhibits in recent years; distributors, service providers, and smaller vendors have taken over.

ITEXPO

I attended several ITEXPO sessions. The most interesting one was "Building the UC Business Case" (Feb 1, 11am) where Irwin Lazar from Nemertes shared results from a recent survey of IT managers about UC deployments. The business case for UC remains elusive. Just 40% of the companies require business case and only 10% measure UC success through cost savings and cost avoidance while the majority relies on user satisfaction (37%), improved collaboration (21%), and feature adoption (19%). When it comes to measuring UC, soft metrics rule the day. Mobility is a key planning concern with 81% of respondents planning to support mobile devices. Finally, Microsoft Lync is making inroads in the enterprise with 19% of respondents deploying it and 37% evaluating it.

My ITEXPO session "Beyond travel avoidance – the real value of HD videoconferencing and collaboration" (Feb 2, 2pm) focused on increasing meeting effectiveness through advanced collaboration capabilities that enhance decision making and improve productivity. The moderator Mark Ricca from IntelliCom Analytics invited three speakers: Scott Morrison, BD Director at Magor Communications, Ron Burns, CEO of ProtonMedia, and me.

My talk focused on the two approaches to improving collaboration and team work with video. The first approach is to improve collaboration capabilities in video solutions - HD content sharing, including to tablets and other mobile devices, white boarding (Polycom UC Board is a good example), studio experience (like in EagleEye Director), and content capturing and management. The second way is to integrate visual communication with collaboration solutions from partners. Good examples are Polycom's integration with Microsoft Lync (for which it was named the 2011 Microsoft UC Innovation Partner of the Year), with IBM SameTime and IBM Connections social business platforms (as announced at LotusSphere in January), and with Jive social media for enterprise.

4GWE Conference 

The 4GWE conference was about building broadband wireless networks with focus on the LTE wireless interface and backhaul technologies. Since I knew a lot about the wireless interface (LTE), I enjoyed the presentation on backhaul technologies by Amir Mekleff, President and CEO, BridgeWay Communications (Feb 1, 1pm).

Due to the trend towards more tablets (tablets outselling laptops) and more smartphones, backhaul bandwidth is and will continue to be bottleneck. Users expect wire-line performance, and are usually disappointed. The sweet spot for LTE networks is microcells with 1-3 miles radius that can deliver up to 100Mbps over the wireless interface or pico cell with 0.1-0.5 miles radius that can deliver up to 300Mbps.

Fiber, cable, and copper are used for backhaul but also increasingly microwave and millimeter wave technologies. While microwaves (6-38GHz spectrum) are good for 4G traffic backhaul over long distances (6GHz can go up to 50 miles, 38GHz can go up to 5 miles), millimeter waves (60-90GHz) are good for 4G traffic backhaul over short distances in densely populated areas. Microwave links have very high penetration in Europe where 60-70% of backhaul via microwave - mostly because old cities are difficult to dig for fiber.

Cloud Communications Summit

I also presented in a session "Can UC in the Cloud?" (Feb 3, 9am) that focused on Unified Communications as a Service. This segment is poised to become nearly a $6 billion dollar market within the next few years. Traditional on-premises solutions will continue to be replaced by more modular and elastic services that can be provisioned, delivered, and monitored through multi-tenant infrastructures, to any user, and location, any device and at any time.

Moderator Thomas Howe, Principal at Embrase, invited three speakers: Davide Petramala, VP Marketing and Sales at Esna Technologies, Chad Krantz, Executive Director Channel Sales at Broadvox, and me. The discussion was mostly about real-time cloud communications, issues with quality of service and what each of participating companies was doing the cloud computing area.

Ingate Summit

Ingate manufactures session border controllers. Since SIP trunking was the most important application for Ingate initially, they started the Ingate SIP Trunking Summit several years ago and always run it in parallel to ITEXPO. They later added UC topics to the Summit and now have 2-2.5 days of educational content. There seems to be demand for education and training because the Summit is very well attended.

I had the chance to present about OVCC at the previous Summit (Austin, September 2011). The hot topic this time was the Ingate Internet+ initiative. The idea goes back to the original SIP architecture (reminds me of discussions 10-12 years ago) that is a flat IP network with end-to-end SIP sessions and IP packets flowing freely end-to-end, too. Unfortunately, voice carriers did not embrace the original SIP voice vision, and created VOIP islands that continue to peer via TDM connections. Since carriers have an established mechanism to trade voice minutes, they never moved pass TDM peering, which in turn means lower voice quality due to multiple conversions of voice from IP to TDM and vice versa; it also creates huge problems for fax. Ingate wants to persuade carriers to peer over IP, so that SIP-based voice, video, IM, presence, etc. can freely flow.

I do not think carriers will rush to embrace the Internet+ idea; the impact on their business model is too significant. However, OVCC managed to bring carriers around the table and reach an agreement on IP peering for video. Carriers seem to see video as very different from voice and do not mind having a different architecture for it. In my view, once the OVCC model has been established for video, we can talk about using the OVCC interconnects to expand the definition with other services: presence, IM, etc. Voice will continue to be the most sensitive topic…

Synopsis Under IP/Patents Telecom Sourcing Conference (SUITS)

I came across SUITS by accident in the lunch break. There was no other free table and I sat next to what it turned out to be a subset of the 20+ corporate attorneys attending the SUITS conference. The official goal of the SUITS event is to advance knowledge innovations of telecommunications, and teach technologists about IPR protection, patent pools, etc.

I had a great conversation with William Geary, Jr. VP of Business Development at MPEGLA who was also one of the key speakers are the event. Bill talked about the efforts in the H.264 licensing pool – Polycom is a part of it – to make H.264 Baseline Profile royalty-free. I really think that makes sense since vendors are moving to higher efficiency profiles such as Main and High. Making the Baseline Profile royalty-free would address some of the issues organizations such as W3C have with adoption of H.264 in their work.

LTE and the Future of Mobile Networking

Upperside Conferences invited me to present about the advances in voice and video technology at the Voice over LTE conference last week. Upperside's conferences always focus on a specific technology subject and perfect if you want to learn "everything" about it. This time the talks were all about LTE, which stands for Long Term Evolution (LTE), and is a key new technology that revolutionizes mobile networks.

LTE versus VoLTE

LTE is a radio access technology that was developed in 3GPP to enhance performance and efficiency in mobile networks, more specifically, increase bit rates (up to 150 Mbps), improve cell spectrum efficiency, and reduce air interface latency. Since LTE itself is a pure IP packet service, transporting voice, SMS, and other media like video and IM has to be specified separately.

Voice over LTE is defined as an end-to-end service that includes not only the efficient LTE radio but also the IP Multimedia Subsystem (IMS) core, the Evolved Packet Core (EPC), and LTE capable devices (dongles, tablets, and smart phones). The key difference between LTE and VoLTE is therefore that LTE is the wireless interface while VoLTE is a complete solution for transporting voice and SMS over the new network.

The problem is that the solution – although fairly new – already needs extension with IM, video, and other functions, and the "voice" in VoLTE is very limiting. So the mobile networking community created something called Real-time Communication Services extension (RCS-e), that extends the voice and SMS services with, for example, IM/chat, file transfer, image and video share. It is roughly the equivalent of what we call Unified Communications in the enterprise communication space. Similar to UC, RCS-e is based on service/capability discovery and uses the SIP protocol. The role of SIP in enterprise UC is described here.

As with any other technology, getting the acronyms right is half of the work, so here are the important ones.

IMS is an architectural framework for delivering multimedia services. The idea is decouple the application from the access method, so that the same functionality can be accesses over wireless (3G, 4G) and fixed networks. This idea is not new: web applications, for example, do not care about the underlying network as long as it carries HTTP; this is probably why they are dubbed Over The Top (OTT) applications in mobile networking lingo. IMS is a revolution in mobile networks where traditionally separate functions were implemented for each type of network. By developing applications only once in the IMS core mobile SPs can shortens time-to-market and compete more successfully with OTT applications. IMS is also designed to provide QOS for applications through the different access networks, and this is a major differentiator for mobile SPs.

EPC is basically an IP router with mobility intelligence that includes handover (switching the connection when the mobile device moves from one radio cell to another), roaming (provides access when users leave their own SP network and enter the network of another SP), etc.

From Circuit Switched Voice to VoLTE

The LTE radio technology is different from 3G and building LTE networks requires substantial capital investment, as pioneers Verizon and MetroPCS in the USA and Vodafone in Europe know very well. As a result, LTE networks will coexist with 2G and 3G networks for long time, and it is critical to find a way to switch calls seamlessly from LTE to non-LTE networks when the user leaves LTE coverage. There are two ways to do that - Single Radio Voice Call Continuity (SRVCC) and Circuit Switched Fall Back (CSFB) – the former using single radio and being less expensive, the latter using dual-radio and costing more.

QOS

In terms of bandwidth, LTE can theoretically provide up to 150Mbps shared in a radio cell. LTE bandwidth is symmetric, that is, upstream bitrate can be equal to downstream bitrate; this makes LTE best choice for symmetric services such as online gaming and real-time voice and video. If there are 200 users in the radio cell, each user can get 0.75Mbps which is enough for high-quality H.264 video.

A bigger QOS problem in LTE networks is network congestion that leads to rapidly increasing delay (and packet loss) with very little advance notice. In video conferencing, the receiving endpoint sends congestion notification to the sender (SIP does that via RTCP while H.323 uses the H.245 Flow Control message), and the sender down-speeds, that is, reduce either resolution or frame rate. Mobile networking vendors are researching ways to detect congestion on a radio cell level; that would require the radio node to send congestion notifications. Since the radio node sees all traffic from all users in the radio cell, it can give an advance warning. It will be important for mobile UC application with video capabilities to listen for such notifications and down-speed – even if their own session is performing well.

Packet loss in LTE networks usually becomes a problem when the user is at the periphery of the cell where the radio signal is weak. When the signal strength is good and the user does not move, frame error rate can be as low as 0.2% which results in negligible packet loss.

To provide Quality of Service to applications, VoLTE defines QOS Class Identifiers (QCI); each of them is appropriate for certain type of traffic. For example, QCI1 bearer is optimized for VoIP/VoLTE. Tests show transmission latency of 140-160ms, which has to be added to the RTP delay and voice/video codec delay. The resulting end-to-end latency can therefore be higher than 200ms, and more work has to be done to reduce the latency below the 200ms limit critical for interactivity on voice and video calls.

Another important QCI for real time communication is QCI5 that is used for signaling (call setup/tear down). Currently, the end-to-end call setup time in 2G/3G networks is about 6 sec, and VoLTE performs better: call setup times measured over the QCI5 bearer are about 2-3 seconds, even when the LTE device is in battery saving mode.

The QOS issues in mobile networks listed above are not very different from the QOS issues in fixed IP networks a decade ago. Video conferencing technology has matured over longer period of time and has therefore already implemented mechanisms for compensating bandwidth reduction (for example, down-speeding implemented in Polycom HDX video endpoints), packet loss (for example, Polycom Lost Packet Recovery), lip sync, etc.

LTE as a Fixed Network Replacement

I often hear that LTE is not a substitute for fixed access networks such as VDSL and FTTH, and my reaction is always "Why not?" As I learned at the VoLTE conference, there is a business case for using LTE instead of DSL to provide high-speed network access. For example, DSL providers in Germany pay a fee of 10 Euros for using the last mile of copper, and Internet SP are very eager to get rid of this cost. Since LTE can provide bandwidths comparable to fixed lines, modem vendors are adding LTE to the portfolio of access technologies. A great example is the FRITZ!Box, the most popular home gateway in Germany, that combines a modem (DSL, cable, and since October - LTE), a router, a firewall, a Wi-Fi access point and a DECT base station. It is a perfect solution for people like me who hate cables lying in the living room or office. Reported throughput over LTE is up to 100Mbps downstream and 50Mbps upstream which makes it Category 3 LTE device.

I have already heard that some service providers in the USA are experimenting with LTE as a fixed line replacement for services to businesses. Considering the cost and time necessary to setup a fixed line (like T1 or T3) to the customer premises, using LTE is a very attractive alternative for service providers. Now imagine that the business has an IP-PBX that uses SIP trunking to connect to a service provider. Bundling LTE and SIP trunking services is suddenly not a far-fetched idea.

All in all, I think LTE will impact both residential high-speed access and business services provided by service providers. The only "if" is Quality of Service. According to the discussion at VoLTE QOS mechanisms in LTE perform well in test environments and initial field deployments but will they do a good job once the LTE network is flooded with LTE capable devices that compete for resources?

Conclusion

The great thing about LTE is that it makes the mobile networks like any other IP network. Enterprise IP applications that used to require complex gateways to interface to mobile networks will now be able to run on the mobile network without much customization. The low hanging fruit is running Unified Communication soft clients like Polycom Real Presence Mobile on media tablets and leveraging the LTE network to connect back to the enterprise network. Going IP end-to-end will help reduce complexity but also cut latency to the absolute minimum. More work will be required in the area of QOS, especially in the congestion detection and notification on a radio cell level.

Mobile Service Providers will continue to develop IMS based application, also leveraging RCS-e, and it will be interesting to track the adoption of IMS applications and other, so called OTT, applications. While most of the Video Networker followers are from enterprise background, and are therefore familiar with the efforts in the enterprise UC environment, we should not ignore the efforts in the mobile networking space to solve the fundamental UC problem.

EduTech and the New Polycom Office in Moscow

The EduTech conference in Moscow was a gathering of representatives from schools, universities, and corporate training organizations in the Russian Federation to discuss new technologies and methods for teaching and training remotely. Understandably, this topic is very hot in a country that stretches over 9 time zones (11 before the reform in 2010) and that requires a lot of communication between Moscow and the regions.

I had the pleasure to present on my favorite topic "Music Performance and Instruction over High-Speed Networks" (in Russian "Видео для музыкального обучения и трансляции концертов"). The presentation was part of a session dedicated to video technology for education, and resulted in many questions and discussions during and after the session. Did I mention that I enjoy presenting in Russian?

While in Moscow, I also got an early peek of the new Polycom office and Executive Briefing Center that officially opened on October 4. The office is located at the Paveletskaya Square which makes it easily accessible from the Domodedovo International Airport (DME) – a non-stop train connects the two - and through the Moscow subway system. The Business Center "Paveletskaya Plaza" is a beautiful 26-story tower visible from afar, and has a modern lobby with well-organized security.

Polycom has the entire 23^rd floor of the building which results in spectacular views in all directions.

The new office is not only home for the Polycom employees in Moscow but also has the latest Polycom technology, including RPX 400 and OTX 300 immersive telepresence systems, all connected via high-bandwidth networks to other Polycom offices. I could not resist the temptation and placed a couple of telepresence calls across the Atlantic. The picture was crystal-clear and stats showed 6Mbps network bandwidth with no packet loss.

Followers of Video Networker in the Russian Federation, I would highly recommend making an appointment (otherwise security will not let you in) and visiting the new Polycom office in Moscow.

How the Migration to IP Improves Voice Quality

Back to the early years of Voice over IP, the quality was not great in comparison to TDM systems. Since IP networks did not have enough bandwidth and quality of service, voice had to be compressed a lot to be sent over the IP network. TDM solutions by contrast did not compress voice and since there was a physical connection between the TDM system and the TDM phone, they did not need to do much packetization either. The result was better voice quality on TDM phones than on IP phones – up until the advance of fast IP LANs and wideband audio codecs in the 2000s entirely changed the balance.

Most VOIP phones shipping today have some sort of HD Voice support. Polycom has been shipping HD voice for 10 years, starting with 7 kHz voice, then moving to 14 kHz audio in 2003 and 20+ kHz audio in 2006. While the voice industry as a whole is only now moving to 7 kHz voice, Polycom has moved further beyond – to support 14 kHz and even 20+ kHz audio (with Siren 22 and G.719 codecs). It is not just about "voice" anymore but rather about "audio" - the technology has gone beyond speech/voice transmission and allows for high-quality music and mixed content.

HD Voice is not only about better quality codecs. The acoustics of the handset were improved while microphones and speakers have to be modified to capture/play higher quality voice and audio. Echo cancelation and other algorithms have to be adjusted to support the wider frequency band. The challenges around transmitting high-quality audio are described in a joint white paper of Polycom and the Manhattan School of Music. The paper highlights our focus on audio quality and demonstrates our capability to meet the requirements of the most demanding users: musicians. The technology developed for this high-end application trickles down to room and personal telepresence systems and telephones, effectively spreading across the entire Polycom portfolio.

So how does a communication system capture the value of high-quality audio now available in VOIP phones? The key is migration to a distributed architecture that routes voice streams without transcoding them back to the TDM format (G.711 codec). If audio is delivered without transcoding between two communication partners on the system, the quality remains the highest (assuming that both partners have high-quality VOIP phones). The matter gets a little more complicated with multipoint calls because most voice conferencing servers embedded in enterprise voice systems support only G.711. If a video conference server such as Polycom RMX is part of a Unified Communication solution, the unused video ports on this server can be configured to support audio (up to the highest audio quality of 20+ kHz). Audio requires far less performance than video; therefore, one video port becomes 40 audio ports, and that is enough scalability for an enterprise deployments. Long-term, however, wideband audio will be gradually supported on all conference servers in enterprise systems, starting with the 7kHz G.722 wideband codec which is widely supported in newer IP phones.

Once the multipoint problem is solved, the only one remaining is connectivity to other systems across service provider networks. Most voice systems today still use TDM connection (such as T1, PRI) to connect to service providers. This TDM connection takes the voice quality down to G.711 due to physical limitations. Newer systems however support the so-called SIP trunking standard (specification is managed by the SIP Forum) that allows connecting the enterprise voice system with a service provider using an IP connection and a virtual trunk with SIP signaling. This virtual trunk does not impose any physical limitations on the voice streams (it is just IP packets crossing the network); therefore, any voice quality can be supported - as long as both the enterprise and the SP systems can handle it. SIP trunks enable wideband voice to travel among enterprise communications systems around the world without any transcoding and quality loss.

Will wideband voice make its way to wireless handsets? The latest generation of wireless handsets – for example Polycom Spectralink 8400 - already support wideband voice (7kHz, G.722, G.722.1), and as long as the voice stream can reach the destination in its original form, the receiver enjoys the clarity and superior understanding of wideband voice communication. The challenges related to multipoint conferencing and SP trunking apply equally to wireless phones, as they are treated as any other phone in the IP communication system environment.

In conclusion, voice technology has made an amazing progress over the past decade. The work of researchers and engineers is now finally finding its way in enterprise communication solutions that provide better quality, reduce misunderstandings and fatigue, and in general, makes human interactions over distances more natural and effortless.

Focus Webinar “The Truth about Unified Communications for SMB”

I got invited to speak about UC and video in a Focus webinar for the Small and Medium Business Community. Since I usually talk to larger organizations, this webinar was a great opportunity to evaluate how the solutions available in the UC and video space apply (or don't apply) to SMBs.

First of all, SMB definition varies by country, for example, the US Government define companies with less than 500 employees as SMBs while in Germany it is companies with less than 250 employees. Since the audience of the webinar was mostly in North America, I created a story around a fictional SMB with about 300 employees distributed across three larger offices and several small sales offices.

The webinar had two parts. In the first 30 minutes, I covered definitions of "collaboration" and "communication", UC scope and market size, and talked about the real value of UC to SMBs. Then I described the types of UC solutions, the value of video as part of the UC solution, and finished by dispelling the myth about superior single-vendor solutions, which also directly relates to the trend towards UC ecosystems, and the increased importance of standards and interoperability.

In the second part, I focused on what SMBs should consider when deploying video. The presentation basically led the audience through the steps of building a video network from scratch to a fully functional multi-site network that spans across geographies and connects to partners, suppliers, and customers. The extensibility and scalability aspect is very important because many SMBs are growing fast and want to make sure a video starter kit can later be expanded to support more users.

I was able to cover not only video fundamentals but also IP network readiness aspects - when connecting distributed offices external organizations.

Webinar attendance was great, and resulted in many questions, some of which I answered in the Q&A session and some online. I am getting a lot of follow-up questions and it looks like the webinar had a huge impact.

If you want to watch the webinar recording, please click here.