Sunday, December 2, 2012

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.

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.

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.

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.

Wednesday, June 6, 2012

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

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

Wednesday, February 22, 2012

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".

Wednesday, February 8, 2012

Cluster of Conferences around ITEXPO


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.


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.