tag:blogger.com,1999:blog-68913597002084938762024-03-13T03:52:26.540-07:00Video NetworkerThis blog discusses collaboration market and technologies including video conferencing, web conferencing, and team collaboration tools. Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.comBlogger63125tag:blogger.com,1999:blog-6891359700208493876.post-37140904712590638252020-10-01T17:15:00.002-07:002020-10-02T10:38:12.680-07:00Twilio SIGNAL Conference <p>The <a href="https://signal.twilio.com/">Twilio SIGNAL</a> virtual
customers and developer conference has just finished and I would like to
capture my impressions while they are still fresh.</p><p class="MsoNormal"><o:p></o:p></p>
<p class="MsoNormal">I wrote about <a href="https://videonetworker.blogspot.com/2012/12/will-webrtc-change-communications_2.html">WebRTC
back in 2012</a> 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. <o:p></o:p></p>
<p class="MsoNormal">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. <a href="https://techcrunch.com/2018/10/15/twilio-acquires-email-api-platform-sendgrid-for-2-billion-in-stock/">Twilio
acquired SendGrid</a>, 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. <o:p></o:p></p>
<p class="MsoNormal">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! <span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><span style="mso-spacerun: yes;"> </span><o:p></o:p></p>
<p class="MsoNormal">Most recently, <a href="https://techcrunch.com/2020/07/09/twilio-acquires-electric-imp-to-bolster-its-growing-iot-business/">Twilio
acquired IoT company Electric Imp</a>, 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 <a href="https://en.wikipedia.org/wiki/ARM_Cortex-M">ARM Cortex-M</a> processor that
allows communication via a lower level encrypted communication channel... back to
Twilio.<o:p></o:p></p>
<p class="MsoNormal">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 <a href="https://www.obamacareusa.org/?;2CPCN_bi3uGVxJDQtzP75HJiVXOGCGdpemoVyzLRkJJnqF9hK1maOtgr7YwoMRxRiYDXDcrh&msclkid=6b143f7514521f2bc022600379d6c44e">Affordable
Care Act (ACA) web site</a> up and running. <span style="mso-spacerun: yes;"> </span>He talked extensively about his approach to
attracting software programmers to work on important government projects such
as Social Security and Veterans Administration. <o:p></o:p></p>
<p class="MsoNormal">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. <span style="mso-spacerun: yes;"> </span><o:p></o:p></p>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-5124980071233875792017-03-10T18:42:00.000-08:002017-03-10T18:46:13.526-08:00Team Collaboration<div class="MsoNormal">
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.</div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
The emerging Team Collaboration market segment is a result
of the collision of two trends:</div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
1) The rise of persistent collaboration spaces to support Agile
and DevOps</div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
2) The rise of messaging in the UCC space </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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…
</div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br />
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. <br />
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<div class="MsoNormal">
<b><i>Conclusion<o:p></o:p></i></b></div>
<br />
<div class="MsoNormal">
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.</div>
Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-19041017263290377152016-11-17T18:04:00.000-08:002016-11-17T19:02:37.546-08:00The Evolution of the Collaboration Room Experience<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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.</div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<div class="MsoNormal">
<br /></div>
<div class="MsoNormal">
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. </div>
<br />
<div class="MsoNormal">
There are good reasons to think hard before making a
decision on collaboration room investments, and I will continue tracking this
topic. </div>
Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-53628215071580819142013-04-10T07:48:00.006-07:002013-04-10T07:49:49.814-07:00EC13 Part 4: Redefining The Conference Room Experience<strong>The new conference room experience is still a work in progress, and the focus is shifting from video to content sharing and annotation.</strong><br />
<strong></strong><br />
This is the last in a four-part series covering some of the major trends I saw at Enterprise Connect Orlando. Friday I covered <a href="http://www.nojitter.com/post/240152271/ec13-report-virtualization" target="_blank">Virtualization</a>, Monday was <a href="http://www.nojitter.com/post/240152300/ec13-report-cloud-services">Cloud Services</a>, and yesterday <a href="http://www.nojitter.com/post/240152318/ec13-report-mobility">Mobility</a>. <br />
<strong></strong><br />
<strong>Redefining the conference room experience</strong><br />
<br />
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 <a href="http://www.microsoft.com/">Microsoft</a> 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. <br />
<br />
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. <br />
<br />
By releasing its Lync Room specification to 4 partners--<a href="http://www.crestron.com/">Crestron</a>, <a href="http://www.lifesize.com/">LifeSize</a>, <a href="http://www.polycom.com/">Polycom</a>, and <a href="http://smarttech.com/">SMART</a>--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. <br />
<br />
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. <br />
<br />
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. <br />
<br />
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 <a href="http://www.ucif.org/">UCIF</a> 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. <br />
<br />
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 <a href="http://www.tely.com/">Tely Labs</a> 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.<br />
<br />
<a href="http://www.revolabs.com/">Revolabs</a> 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.<br />
<strong></strong><br />
<strong>Conclusion</strong><br />
<br />
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. Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com1tag:blogger.com,1999:blog-6891359700208493876.post-83839265406916385592013-04-09T10:56:00.006-07:002013-04-09T10:58:21.046-07:00EC13 Report Part 3: Mobility<strong>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.</strong><br />
<br />
This is the third in a four-part series covering some of the major trends I saw at Enterprise Connect Orlando. Friday I covered <a href="http://www.nojitter.com/post/240152271/ec13-report-virtualization">Virtualization</a>, and yesterday <a href="http://www.nojitter.com/post/240152300/ec13-report-cloud-services">Cloud Services</a>; tomorrow's topic will be the Conference Room Experience.<br />
<br />
<strong>Mobility</strong><br />
<br />
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. <br />
<br />
Extensions to the call processing software allow switching (or shifting) sessions across any number of devices associated with a user. <a href="http://www2.alcatel-lucent.com/enterprise-and-industries/">Alcatel-Lucent</a>'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.<br />
<br />
<a href="http://www.thrupoint.com/">Thrupoint</a> 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.<br />
<br />
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.<br />
<br />
<a href="http://www.aastrausa.com/">Aastra</a> 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. <br />
<br />
<a href="http://www.shoretel.com/">ShoreTel</a> 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. <strong></strong><br />
<strong></strong><br />
<strong>Conclusion</strong><br />
<br />
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. Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-33857639868108927862013-04-08T15:48:00.003-07:002013-04-08T15:54:55.774-07:00EC13 Report Part 2: Cloud Services<br />
<div class="text">
<div style="display: none;">
<!-- Add the following three tags to your body for Google+ Button --><span itemprop="name">EC13 Report: Cloud Services</span><span itemprop="description">Vendors are rushing into creating cloud services to reach new customers--but there are concerns to address.</span></div>
<strong>Vendors are rushing into creating cloud services to reach new customers--but there are concerns to address.</strong> <br />
<br />
This is the second in a four-part series covering some of the major trends I saw at Enterprise Connect Orlando. <a href="http://www.nojitter.com/post/240152271/ec13-report-virtualization">On Friday</a> I covered Virtualization; tomorrow's topic will be Mobility, and Wednesday's will be the Conference Room Experience.<strong></strong><br />
<br />
<strong>Cloud Services</strong><br />
<br />
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 <a href="http://www.avispl.com/">AVI-SPL</a>, I met with <a href="http://www.appliedglobal.com/">AGT</a>, which differentiates itself through offering software-based MCU in the cloud and offering management of video endpoints. <br />
<br />
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.<br />
<br />
The cloud allows some traditional on-premise vendors to address very small customers that cannot really afford an on-premise solution. <a href="http://www.inin.com/">Interactive Intelligence</a> 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 <a href="http://www.five9.com/">Five 9</a>, a pure-play cloud call center offering. <br />
<br />
On the video side, <a href="http://www.magorcorp.com/">Magor</a> 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.<br />
<br />
Since these offerings rely on Internet best effort service, some industry analysts warn against it, while others embrace the Internet quality. <a href="http://www.unifiedoffice.com/">Unified Office</a> 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.<br />
<br />
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. <br />
<br />
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. <a href="http://www.broadsoft.com/">BroadSoft's</a> definition, for example, is having BroadWorks system in the SP data center while getting UC services from the BroadCloud. <br />
<br />
<strong>Conclusion</strong><br />
<br />
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. </div>
Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-75776320194017312532013-04-04T22:01:00.003-07:002013-04-08T15:50:42.327-07:00EC13 Report Part 1: Virtualization
<br />
<div class="text" style="display: none;">
<!-- Add the following three tags to your body for Google+ Button --><span itemprop="name">EC13 Report: Virtualization</span><span itemprop="description">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.</span></div>
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<strong>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.</strong> </div>
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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.</div>
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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 <strong>virtualization, cloud services, mobility,</strong> and <strong>redefining of the conference room experience</strong>. I have written a post on each of these topics, and these will run on <a href="http://www.nojitter.com/post/240152271/ec13-report-virtualization" target="_blank">No Jitter</a> over the next 4 days, beginning with virtualization.</div>
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<strong>Virtualization</strong></div>
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<a href="http://www.nojitter.com/post/240147929/virtualization-of-communication-infrastructure" target="_blank">Virtualization</a>--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:</div>
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* Trailblazer <a href="http://www.mitel.com/">Mitel</a> 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. </div>
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* <a href="http://www.avaya.com/">Avaya</a> has virtualized its Avaya Communications Manager and Contact Center application as well as several other elements of the Aura architecture. </div>
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* <a href="http://www.shoretel.com/">ShoreTel</a> 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. </div>
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* <a href="http://www2.alcatel-lucent.com/enterprise-and-industries/">Alcatel-Lucent</a> 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. </div>
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* <a href="http://www.siemens-enterprise.com/us/">Siemens Enterprise Communications</a> is in the process of certifying its OpenScape portfolio with VMware. * <a href="http://www.necam.com/">NEC</a> has already virtualized UNIVERGE 3C, its Windows-based UC system from the Sphere acquisition. </div>
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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. </div>
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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. <a href="http://www.aastrausa.com/">Aastra</a> 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. </div>
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In the video conferencing world, <a href="http://www.polycom.com/">Polycom</a> 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. </div>
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<a href="http://www.avaya.com/">Radvision</a> (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.</div>
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<a href="http://www.avispl.com/">AVI-SPL</a> 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.</div>
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Looking across the contact center industry to test the hypothesis that virtualization is an universal trend, I came across <a href="http://www.five9.com/">Five9</a>, a cloud call center provider that uses virtualization in its network of leased data centers, and <a href="http://calabrio.com/">Calabrio</a>, 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. </div>
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In the voice networking segment, Session Border Controller (SBC) vendors are beginning to think about virtualization, with <a href="http://www.sonus.com/">Sonus</a> 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. </div>
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I was not sure I would find virtualization in <a href="http://www.adtran.com/">Adtran</a>'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. </div>
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Virtualization also found its way in <a href="http://www.extremenetworks.com/">Extreme Networks'</a> 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. </div>
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<strong>Conclusion</strong></div>
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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. </div>
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Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-42561114335116043802013-02-18T10:08:00.002-08:002013-02-18T10:13:10.192-08:00Controlling the BordersAfter Oracle announced the <a href="http://www.oracle.com/us/corporate/press/1903221">acquisition of Acme Packet</a>, 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.<br />
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<u>SBC Origin</u> <br />
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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. <br />
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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. <br />
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Over the years, I have attended a lot of <a href="http://www.ietf.org/">IETF</a> 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 <a href="http://www.itu.int/">ITU </a>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 <a href="http://www.itu.int/rec/T-REC-H.460.19/en">ITU-T H.460</a> standard. <br />
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<u>SBC Functionality</u><br />
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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. <br />
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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.) <br />
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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).<br />
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<u>SBC Market Dynamic</u><br />
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<a href="http://www.acmepacket.com/">Acme Packet</a> 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 <a href="http://www.ericsson.com/">Ericsson</a> 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 <a href="http://www.juniper.net/us/en/company/press-center/press-releases/2005/kgr/pr-050329.html">Kagoor</a> in 2005. Ditech Communications acquired <a href="http://en.wikipedia.org/wiki/Jasomi_Networks">Jasomi</a> in 2005 (Nuance Communications acquired Ditech in 2012). Nextone was swallowed by Nextpoint which <a href="http://www.genband.com/">Genband</a> acquired in 2008. <a href="http://www.audiocodes.com/">AudioCodes</a> acquired Netrake in 2006. <br />
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Big VOIP vendors took various approaches to adding SBC functionality to their portfolios. <a href="http://www.cisco.com/">Cisco</a> developed its own Cisco Unified Border Element (CUBE) while <a href="http://www.avaya.com/">Avaya</a> acquired <a href="http://www.avaya.com/usa/about-avaya/newsroom/news-releases/2011/pr-111003">Sipera</a> in 2011. In the service provider market, <a href="http://www.broadsoft.com/">BroadSoft</a> relied on Acme Packet (in fact that is how I first came across Acme Packet years ago) while others like <a href="http://www.8x8.com/">8x8</a> developed their own SBC technology. <br />
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The SBC market attracted competition. In 2010, <a href="http://www.sonus.net/">Sonus Networks</a> 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 <a href="http://www.sonus.net/net/pdfs/SONS_News_2012_6_19_General_Releases.pdf">acquired Network Equipment Technologies (NET)</a> 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. <a href="http://www.sansay.com/">Sansay</a>, the #3 in SBC market, emerged from the former Nuera Communications team in San Diego in 2002. At ITexpo 2013, Sansay won the <a href="http://sip-trunking.tmcnet.com/news/2013/02/07/6910658.htm">“Best Service Provider Solution”</a> award for their transcoding technology, integrated into the SBC product.<br />
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Through the years, I got to know two other players in the SBC market. <a href="http://edgewaternetworks.com/">Edgewater Networks</a> 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. <a href="http://www.ingate.com/">Ingate Systems</a> focuses on SIP trunking and hosts its <a href="http://itexpo.tmcnet.com/west/collocated-event/w11-ingates-sip-trunking-workshop.htm">SIP Trunking Summit / Workshop</a> at every IT Expo event; this event evolved into the <a href="http://www.ingate.com/itexpo_miami_2013.php">SIP Trunking - UC Seminars</a>. I had the pleasure to present at several Ingate events, and can say that they are very well-organized, well-attended, and very educational. <br />
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<u>SBC’ Role in the Enterprise</u><br />
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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.<br />
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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. <br />
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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. Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com1tag:blogger.com,1999:blog-6891359700208493876.post-4740032350961538842012-12-02T22:15:00.001-08:002013-01-25T10:37:46.697-08:00Will WebRTC Change The Communications Industry?Back in April 2011, I wrote about the WebRTC discussion at <a href="http://videonetworker.blogspot.com/2011/04/what-did-80th-meeting-of-internet.html" target="_blank">the 80th IETF meeting</a> 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 <a href="http://www.webrtcworld.com/conference/" target="_blank">WebRTC Conference and Expo</a> in San Francisco that gathered about <a href="http://www.flickr.com/photos/20518315@N00/8236143191/in/photostream" target="_blank">300 participants</a>, and focused on two questions: a) is the WebRTC technology ready for deployment and b) what is its impact on the communications industry? <br />
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<u>The Surprise Sponsor</u> <br />
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.<br />
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<u>WebRTC Vision And Realty</u> <br />
WebRTC’s vision is to provide interoperable royalty-free high-quality communication between browsers. To deliver on this vision, <a href="http://googlepress.blogspot.com/2009/08/google-to-acquire-on2-technologies_05.html" target="_blank">Google acquired On2 Technologies</a> 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. <br />
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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 <a href="http://docs.polycom.com/global/documents/whitepapers/g719-the-first-itut-standard-for-full-band-audio.pdf">G.719</a> used in video conferencing systems, having Opus audio in the web browser will definitely drive web developers towards using it in voice/video applications. <br />
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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. <br />
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<u>WebRTC Status</u><br />
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. <br />
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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.<br />
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<u>Mobile</u><br />
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. <br />
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In mobile networks, there are several new communication methods that are competing for attention and resources. Similar to WebRTC, <a href="http://videonetworker.blogspot.com/2011/11/upperside-conferences-invited-me-to.html" target="_blank">Voice over LTE</a> 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 <a href="http://www.crocodile-rcs.com/" target="_blank">Crocodile</a>. <br />
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<u>Demos</u> <br />
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.<br />
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<u>Industry Impact</u><br />
The current consensus is that while WebRTC clients will become very cheap or free there will be business opportunities around infrastructure and services.<br />
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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.<br />
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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. <br />
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<u>Takeaways</u><br />
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). <br />
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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.<br />
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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.<br />
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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. <br />
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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.Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-33848346664561062062012-06-06T16:42:00.002-07:002012-06-06T16:42:17.653-07:00New 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. <br />
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. <br />
The paper is available to WR subscribers at <a href="http://wainhouse.com/index.php">http://wainhouse.com/index.php</a>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com2tag:blogger.com,1999:blog-6891359700208493876.post-51985165673477928042012-06-06T16:32:00.001-07:002012-06-06T16:32:40.856-07:00New 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. <br />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. <br />The paper is available to WR subscribers at <a href="http://wainhouse.com/index.php">http://wainhouse.com/index.php</a>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-52530418843774823312012-02-22T10:48:00.001-08:002013-01-28T15:24:33.304-08:00Ubiquitous Visual Communications at PTC 2012<span xmlns=""></span><br />
<a href="http://www.ptc.org/ptc12/?page_id=6"><span style="font-family: Arial, Helvetica, sans-serif;">The </span><span style="font-family: Arial, Helvetica, sans-serif;">34th<span style="font-size: small;"> PTC Conference</span></span></a><span style="font-family: Arial, Helvetica, sans-serif;"><span style="font-family: Arial, Helvetica, sans-serif;"> (tagline</span> "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 </span><a href="http://www.ptc.org/ptc11/index.php?page_id=14&seshid=314"><span style="font-family: Arial, Helvetica, sans-serif;">Unified Communications Services</span></a><span style="font-family: Arial, Helvetica, sans-serif;">, this year's topic was (Video) </span><a href="http://www.ptc.org/ptc12/index.php?page_id=31&seshid=464"><span style="font-family: Arial, Helvetica, sans-serif;">Interoperability, Interconnection, and Sky-Rocketing Global Utility</span></a><span style="font-family: Arial, Helvetica, sans-serif;">. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">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 </span><a href="http://www.ptc.org/ptc12/index.php?page_id=31&seshid=464"><span style="font-family: Arial, Helvetica, sans-serif;">prominently featured on the PTC web site</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> and now includes the slides from the three presentations. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">The session included </span><a href="http://www.flickr.com/photos/20518315@N00/6758518185/"><span style="font-family: Arial, Helvetica, sans-serif;">three speakers</span></a><span style="font-family: Arial, Helvetica, sans-serif;">: 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. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">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. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">Damian's presentation generated questions from other service providers that consider joining OVCC or are in the process of joining OVCC. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">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.</span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">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. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">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 </span><a href="http://www.polycom.com/global/documents/whitepapers/polycom-enhances-portfolio-with-support-of-tip.pdf"><span style="font-family: Arial, Helvetica, sans-serif;">TIP</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> 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). </span><br />
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<span style="font-family: Times New Roman;"><span style="font-family: Arial, Helvetica, sans-serif;">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".</span> </span>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-56039183839999684182012-02-08T20:16:00.001-08:002012-02-09T21:27:23.786-08:00Cluster of Conferences around ITEXPO<span xmlns=""></span><br />
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<span style="font-size: small;">Summary</span></h1>
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.<br />
The <a href="http://itexpo.tmcnet.com/east12/attendees/e12-conferences-program.aspx">conference part</a> 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. <br />
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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. <br />
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Key analysts in the industry attend ITEXPO to moderate sessions, present, meet with vendors, and get updates on their solutions. <br />
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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. <br />
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<span style="font-size: small;">ITEXPO</span></h1>
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. <br />
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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. <br />
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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.<br />
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<strong>4GWE Conference </strong><br />
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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).<br />
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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. <br />
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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. <br />
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<span style="font-size: small;">Cloud Communications Summit</span></h1>
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.<br />
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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. <br />
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<span style="font-size: small;">Ingate Summit </span></h1>
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. <br />
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I had the chance to present about <a href="http://ovcc.net/">OVCC</a> 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.<br />
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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…<br />
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<span style="font-size: small;">Synopsis Under IP/Patents Telecom Sourcing Conference (SUITS) </span></h1>
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.<br />
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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. <br />
<br />Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-10409849711287052472011-11-19T14:59:00.001-08:002011-11-19T14:59:01.000-08:00LTE and the Future of Mobile Networking<span xmlns=''><p><span style='font-family:Arial'><br/>Upperside Conferences invited me to present about the advances in voice and video technology at the <a href='http://www.uppersideconferences.com/volte2011/volte2011program.html'>Voice over LTE conference</a> 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.<br/><br/><span style='text-decoration:underline'>LTE versus VoLTE</span><br /> </span></p><p><span style='font-family:Arial'>LTE is a radio access technology that was developed in <a href='http://www.3gpp.org/'>3GPP</a> 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. <br /></span></p><p><span style='font-family:Arial'>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. <br /></span></p><p><span style='font-family:Arial'>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 <a href='http://www.polycom.com/global/documents/whitepapers/unified_communications_drive_protocol_convergence.pdf'>here</a>. <br /></span></p><p><span style='font-family:Arial'>As with any other technology, getting the acronyms right is half of the work, so here are the important ones.<br /></span></p><p><span style='font-family:Arial'>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.<br /></span></p><p><span style='font-family:Arial'>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.<br/><br/><span style='text-decoration:underline'>From Circuit Switched Voice to VoLTE</span><br /> </span></p><p><span style='font-family:Arial'>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.<br/><br/><span style='text-decoration:underline'>QOS </span><br /> </span></p><p><span style='font-family:Arial'>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. <br /></span></p><p><span style='font-family:Arial'>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. <br /></span></p><p><span style='font-family:Arial'>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.<br /></span></p><p><span style='font-family:Arial'>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. <br /></span></p><p><span style='font-family:Arial'>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. <br /></span></p><p><span style='font-family:Arial'>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. <br/><br/><span style='text-decoration:underline'>LTE as a Fixed Network Replacement</span><br /> </span></p><p><span style='font-family:Arial'>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. <br /></span></p><p><span style='font-family:Arial'>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. <br /></span></p><p><span style='font-family:Arial'>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?<br/><br/><span style='text-decoration:underline'>Conclusion</span><br /> </span></p><p><span style='font-family:Arial'>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. <br /></span></p><p><span style='font-family:Arial'>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. </span></p></span>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-73387859580894911562011-10-06T17:34:00.001-07:002011-10-06T17:34:36.195-07:00EduTech and the New Polycom Office in Moscow<span xmlns=''><p>The <a href='http://www.flickr.com/photos/20518315@N00/6191211683/'>EduTech conference</a> 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. <br /></p><p>I had the pleasure to present on my favorite topic <a href='http://www.polycom.com/global/documents/whitepapers/music_performance_and_instruction_over_highspeed_networks.pdf'>"Music Performance and Instruction over High-Speed Networks"</a> (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?<br /></p><p>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 <a href='http://www.flickr.com/photos/20518315@N00/6191681132/in/photostream'>Business Center "Paveletskaya Plaza</a>" is a beautiful 26-story tower visible from afar, and has a <a href='http://www.flickr.com/photos/20518315@N00/6191165127/in/photostream'>modern lobby</a> with well-organized security. <br /></p><p>Polycom has <a href='http://www.flickr.com/photos/20518315@N00/6191166149/in/photostream'>the entire 23<sup>rd</sup> floor</a> of the building which results in <a href='http://www.flickr.com/photos/20518315@N00/6191168045/in/photostream'>spectacular views</a> in <a href='http://www.flickr.com/photos/20518315@N00/6191686678/in/photostream'>all directions</a>. <br /></p><p><a href='http://www.flickr.com/photos/20518315@N00/6191684086/in/photostream'>The new office</a> is not only home for the Polycom employees in Moscow but also has the latest <a href='http://www.flickr.com/photos/20518315@N00/6218981172/in/photostream'>Polycom technology</a>, including RPX 400 and <a href='http://www.flickr.com/photos/20518315@N00/6218981402/in/photostream'>OTX 300</a> 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. <br /></p><p>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.<br /></p></span>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-87752584756643454752011-09-12T08:08:00.001-07:002011-09-12T08:08:41.182-07:00How the Migration to IP Improves Voice Quality<span xmlns=''><p>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. <br /></p><p>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. <br /></p><p>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 <a href='http://www.polycom.com/global/documents/whitepapers/music_performance_and_instruction_over_highspeed_networks.pdf'>joint white paper of Polycom and the Manhattan School of Music</a>. 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.<br /></p><p>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.<br /></p><p>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 <a href='http://www.sipforum.org/sipconnect'>SIP Forum</a>) 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. <br /></p><p>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.<br /></p><p>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. </p></span>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com1tag:blogger.com,1999:blog-6891359700208493876.post-33379788727450183952011-07-14T12:59:00.001-07:002011-07-14T12:59:01.867-07:00Focus Webinar “The Truth about Unified Communications for SMB”<span xmlns=''><p><span style='font-family:Arial; font-size:10pt'>I got invited to speak about UC and video in a <a href='https://vts.inxpo.com/Launch/Event.htm?ShowKey=5869'>Focus webinar for the Small and Medium Business Community</a>. 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. <br /></span></p><p><span style='font-family:Arial; font-size:10pt'>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. <br /></span></p><p><span style='font-family:Arial; font-size:10pt'>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.<br /></span></p><p><span style='font-family:Arial; font-size:10pt'>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. <strong><br /> </strong></span></p><p><span style='font-family:Arial; font-size:10pt'>I was able to cover not only video fundamentals but also IP network readiness aspects - when connecting distributed offices external organizations. <br /></span></p><p><span style='font-family:Arial; font-size:10pt'>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. <em><br /> </em></span></p><p><span style='font-family:Arial; font-size:10pt'>If you want to watch the webinar recording, please click <a href='https://vts.inxpo.com/Launch/Event.htm?ShowKey=5869'>here</a>. <br /></span></p></span>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-9852833172608593842011-06-13T16:23:00.000-07:002011-06-13T16:23:30.055-07:00“Music Performance and Instruction over High Speed Networks”I have written many white papers but the one that elicits the strongest emotional response has always been “Music Performance and Instruction over High Speed Networks” which Christianne Orto and I wrote back in 2008. This paper tells the fascinating story of collaboration between Polycom and the Manhattan School of Music to enable remote music performances and instructions.<br />
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A lot of things have happened since 2008, for example, Polycom introduced new technologies in the area of voice and video communication while MSM found new applications for the technology. Few weeks ago, Christianne and I talked about the need to update the paper in early June, just before the <a href="http://www.iste.org/conference.aspx">International Society for Technology in Education (ITSE) conference</a> in Philadelphia and the <a href="http://www.terena.org/activities/network-arts/barcelona/">Network Performing Arts Production workshop</a> in Barcelona. The updated white paper has just been posted <a href="http://www.polycom.com/global/documents/whitepapers/music_performance_and_instruction_over_highspeed_networks.pdf">here</a>. Please have a look and send us your feedback!Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-81507921566925497702011-06-10T14:52:00.000-07:002011-06-10T14:53:59.540-07:00How Will the Migration from IPv4 to IPv6 Impact Voice and Visual Communication? Take 2<div class="MsoNormal" style="margin: 0in 0in 10pt;"><a href="http://www.blogger.com/" name="OLE_LINK2"></a><span style="mso-bookmark: OLE_LINK2;"><span style="font-family: "Arial", "sans-serif";">The “World IPv6 Day” (June 8, 2011) was the first global test intended to help service providers and vendors prepare for the inevitable migration to IPv6. </span></span><span style="font-family: "Arial", "sans-serif";">How is IPv6 different from IPv4? Why is IPv6 so important to the Internet and private intranets? What is driving IPv6 adoption? How will the migration to IPv6 affect voice and visual communication? Is Polycom ready for IPv6? Find answers in <a href="http://www.polycom.com/global/documents/whitepapers/ipv4-to-ipv6-migration-whietpaper.pdf"><span style="color: purple;">my new white paper</span></a>. </span></div>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com1tag:blogger.com,1999:blog-6891359700208493876.post-10387317342016208062011-05-06T17:47:00.001-07:002011-05-06T18:13:45.347-07:00What is New in the US Research and Education Community?<span xmlns=""></span><br />
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<a href="http://www.internet2.edu/"><span style="font-family: Arial;">Internet2</span></a><span style="font-family: Arial;"> is a non-profit organization that operates the high-speed backbone for the US Research and Education (R&E) community. It counts 200+ of the largest US universities and research organizations as members plus a lot of other members - international partners, vendors, etc. - to a total of about 350 members. I have represented Polycom in Internet2 since 2007, and sit on one of the governing councils called Application, Middleware, and Services Advisory Council, or <a href="https://wiki.internet2.edu/confluence/display/I2AC/Roster+of+AMSAC+Members">AMSAC</a>. </span><span style="font-family: Arial;">Internet2 members meet twice a year. While the Fall Internet2 Member Meeting moves around the country (next one will be in Raleigh, NC), the spring event is always in Arlington, Virginia. <a href="http://events.internet2.edu/2011/spring-mm/">The latest meeting</a> took place April 18-20, 2011, and was another great opportunity to meet the US R&E community and some international participants. </span><br />
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<span style="font-family: Arial;">Here are some meeting highlights:</span><br />
<ul><li><span style="font-family: Arial;">Internet2 CEO David Lambert announced the new Internet2 initiative in cloud services and the new Network Development and Deployment Initiative (NDDI).</span></li>
<li><span style="font-family: Arial;">Internet2 is expanding its high-speed backbone network</span></li>
<li><span style="font-family: Arial;">Video is a hot topic for the US R&E community</span></li>
<li><span style="font-family: Arial;">There is a need for new audio-video infrastructure to connect the R&E community</span></li>
<li><span style="font-family: Arial;">Migration from IPv4 to IPv6 may not be an issue in the backbone anymore but local R&E networks are still struggling, as are some commercial providers</span></li>
<li><span style="font-family: Arial;">Wide deployment of digital certificates in the R&E community improves network security </span> </li>
</ul><br />
<span style="font-family: Arial; text-decoration: underline;">New Internet2 Initiatives </span><br />
<br />
<span style="font-family: Arial;">David Lambert, Internet2 CEO, announced that I2 and HP were working on cloud services. Internet2 has spent quite a lot of time looking for the appropriate partner in this space and the HP offer was best suited for the needs of the R&E community. </span><span style="font-family: Arial;">David announced the Network Development and Deployment Initiative (NDDI) that includes I2, Indiana University, and Stanford University (Clean Slate Program). Internet2 will offer a new service - Open Science Scholarship and Science Exchange (OS3E) - to meet community requirements. The service will be first available in fall 2011 and will use <a href="http://www.openflow.org/">OpenFlow</a> technology. The goal is to create the equivalent of Linux for networking and allow for open source development. They basically asked switch/router vendors to turn off the control plane and allow remote computers to control them. Internet2 will be a national test bed for OpenFlow. Matt Davy from the Global Research NOC at Indiana University and Rob Vietzke from Internet2 will lead the project. They will work closely with international partners: CANARIE (Canada), JANET (UK), GEANT (Europe), JGNX (Japan), and RNP (Brazil).</span><br />
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<span style="font-family: Arial;">How does that relate to video communications? There have been efforts in the industry to make the IP networks video application aware, and that requires communication between a call control engine on the application side and a policy engine on the IP networking side. The limitation is that each IP networking vendor uses a different policy engine, and there is no single application that can control the entire mixed-vendor network. With the new OpenFlow architecture there is a "standard" API to talk to all IP networking equipment, no matter who makes it. That will potentially give us even more control of the end-to-end QOS in the IP network, which is a benefit to video applications.</span><br />
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<span style="font-family: Arial; text-decoration: underline;">Internet2 is Expanding its High-speed Backbone Network </span><br />
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<span style="font-family: Arial;">US UCAN funding will be used to expand the Internet2 network. <a href="http://www.flickr.com/photos/20518315@N00/5693834191/in/photostream/">Map of the network expansion</a> was presented in the demo area. The middle section connecting West and East Coasts will be built first, followed by the south span, then the north span of the network. The expansion will require building several new so Giga-PoPs that host optical and IP routing equipment. I took a picture of the equipment that is installed in such <a href="http://www.flickr.com/photos/20518315@N00/5694414234/in/photostream">GigaPoP</a>. On the left side is the Ciena optical equipment. On the right side are a small Cisco 2600 router, an HP server, and a giant Juniper T1600 router with huge blades. </span><span style="font-family: Arial;">The expansion of the Internet2 backbone is necessary to carry the additional traffic from anchor institutions: community centers, rural hospitals, etc. Applications such as distance learning and telehealth will drive video traffic from and to these institutions, and result in a lot of video traffic over the expanded Interent2 backbone. </span><span style="font-family: Arial;"><br />
</span><span style="font-family: Arial; text-decoration: underline;"></span><br />
<span style="font-family: Arial; text-decoration: underline;">Video is a Hot Topic for the US R&E Community</span><br />
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<span style="font-family: Arial;">The session <a href="http://events.internet2.edu/2011/spring-mm/agenda.cfm?go=session&id=10001703&event=1035">"Where Videoconferencing and Telepresence Meet Immersion and Interoperability"</a> drew a lot of attention. Internet2 members are big video users and Internet2 itself offers video services to the community. Polycom has been partnering with Intrenet2 for many years and a lot of the services are leveraging Polycom infrastructure. </span><span style="font-family: Arial;">Ben Fineman from Internet2, talked about a successful telepresence interop test with 32 telepresence screens, connecting equipment from Polycom, Cisco, LifeSize, etc. My presentation focused on <a href="http://videonetworker.blogspot.com/2009/10/telepresence-interoperability.html">telepresence interop</a> and the challenges of connecting multi-screen (multi-codec) systems. I provided an overview of the Telepresence Interoperability Protocol (TIP) that Polycom will be supporting within few months to enable short-term interop across Polycom and Cisco telepresence systems. Then I focused on the long-term telepresence interoperability efforts in the IETF CLUE Working Group, and on Polycom's work in this area. Since I attended <a href="http://videonetworker.blogspot.com/2011/04/what-did-80th-meeting-of-internet.html">the last IETF meeting</a>, I was able to provide a lot of detail about CLUE and answer questions from the audience. </span><br />
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<span style="font-family: Arial;">The third presenter in the session was supposed to be Sean Lessman from Cisco but meeting he canceled right before the since he was on his way out of Cisco. In the last minute Michael Harttree from the Cisco CTO office jumped in. Michael was not very familiar with telepresence and talked instead about the trend towards more video (streaming, surveillance, etc.) in the network. There are many types of video floating around and the challenge is how to separate them and treat them appropriately (in terms of latency budget) on the IP network. This reminded me of the discussion in the IETF MMUSIC group about more detailed description of the type of traffic in SDP so that this description can be preserved across SP networks (which modify QOS settings).</span><br />
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<span style="font-family: Arial; text-decoration: underline;">New Infrastructure for Audio and Video Services to the R&E Community</span><br />
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<span style="font-family: Arial;">I have been attending meetings of the Audio Video Communication Infrastructure Special Interest Group (SIG) for quite a while. The group focuses on connecting VOIP and video networks with PBX and PSTN to deliver seamless communication across the R&E community. Hot topic is the use of E.164 numbers versus alternatives such as SIP URIs, leveraging standards such as <a href="http://www.enum.org/what">ENUM</a> and existing systems such as <a href="http://en.wikipedia.org/wiki/Global_Dialing_Scheme">GDS</a>. </span><br />
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<span style="font-family: Arial;">Ben Fineman from Internet2 and Walt Magnussen from Texas A&M are very active in this group, and I always enjoy the opportunity to discuss with them. The consensus so far is that Internet2 should request from ITU an international "country code" that would allow Internet2 to assign numbers across the R&E community. Agreement with commercial SPs have to be signed to make sure the traffic is routed appropriately. I am very excited about that topic because a lot of new Unified Communications services can be developed for the R&E community on the IP network. (Unfortunately,) the connectivity to PSTN is still essential for the success of UC deployments. </span><br />
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<span style="font-family: Arial; text-decoration: underline;">Migration from IPv4 to IPv6</span><br />
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<span style="font-family: Arial;">Leslie Daigle from the Internet Society (ISOC) delivered a keynote about the importance of IPv6. Only a very small portion of Internet traffic today is IPv6, and businesses have claimed for long time that there is no business case for IPv6. On the other hand, the need for IP address space is big, and companies are trying to buy address space from other users. Based on the Avaya-Nortel acquisition, we know that the price for an IPv4 address is $11.25. But residential and mobile providers need even bigger IP address space than enterprise. Content providers also need to enable IPv6 in their services. IPv6 is gradually starting to make business sense because IPv4 addresses have price attached, NATs are hard and expensive, certain apps, e.g. games, do not work well in NAT environment.</span><br />
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<span style="font-family: Arial;">When Google turned on IPv6 on YouTube, IPv6 traffic spiked. That means there are a lot of IPv6 clients out there. It is estimated that about 0.5% of Google customers will not be able to reach the service if Google alone turns on IPv6. They do not want to lose customers to others; therefore, Google, Yahoo, and Facebook agreed to turn on IPv6 for 24 hours on June 8, 2011 (<a href="http://isoc.org/wp/worldipv6day/">World IPv6 Day</a>). Note that IPv4 will not be turned off. </span><br />
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<span style="font-family: Arial;">The call to action to service providers is to announce plans for IPv6 and create momentum around it. It is important for network administrators to include the World IPv6 Day in their change plan, so that no other changes happen on that exact day. </span><span style="font-family: Arial;">With all of the excitement around the migration to IPv6, I decided to write a white paper on that issue. I have tons of information about IPv6 (some is captured in a <a href="http://videonetworker.blogspot.com/2009/05/how-will-migration-from-ipv4-to-ipv6.html">previous post</a>) and intend to focus on the impact of IPv6 on video communications. Stay tuned! I will post a link to the paper when it is ready.</span><br />
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<span style="font-family: Arial; text-decoration: underline;">Digital Certificates to Improve Network Security in the R&E Community </span><br />
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<a href="http://www.incommonfederation.org/"><span style="font-family: Arial;">InCommon</span></a><span style="font-family: Arial;"> is a part of Internet2 that provides services to the R&E community. These services range from authentication to group management to – recently – low-cost digital certificates. Security is very important for voice and video communications, and Polycom products support digital certificates, so I was curious how universities deploy them. </span><br />
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<span style="font-family: Arial;">John Krienke, Internet2 COO, talked about the partnership between Internet2 and <a href="http://www.comodo.com/resources/small-business/digital-certificates-intro.php">Comodo</a>. Comodo listened to the requirements for campus administration, and allows sub-domains for local certificate management. They provide tools to find all server certificates, and since the Comodo license is per site, you can assign a certificate to each server and do not need wildcard certificates. </span><br />
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<span style="font-family: Arial;">Paul Kaski from the University of Texas System shared his experience with the InCommon certificate service. His organization used VeriSign for 11 years but due to budgetary constraints could not afford the steep price tag anymore and started using the InCommon service in 2H'2010. The estimated cost saving is $325K per year. The main advantages of the InCommon service are very quick SSL certificates approval, easy admin interface, and available API for both SSL and user certificates.</span><br />
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<span style="font-family: Arial;">Digital certificates are a great way to authenticate users, devices, and servers in the network. Certificates definitely increase security in the network, and the only drawback I can think of is the cost. Now that R&E organizations have access to lower cost certificates and to the tools to manage them in campus environment, I expect wide adoption.</span><br />
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<span style="font-family: Arial; text-decoration: underline;">Conclusion</span><br />
<br />
<span style="font-family: Arial;">The Spring Interenet2 Member Meeting was a great opportunity to take a snapshot of the technology developments in the US R&E community. It is ahead of the commercial sector in some areas (advanced networking, IPv6 migration) and lagging is others (applications). I think there is an opportunity for commercial vendors, especially the ones like Polycom who rely on open standards and interoperability, to participate in the creation of new applications and services for the R&E community. </span>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-27667912130515167492011-04-15T17:11:00.001-07:002011-04-29T11:51:29.226-07:00Unified Communications Forum in Moscow<span xmlns=""></span><br />
<span style="font-family: Arial;">The <a href="http://www.unicomm-forum.ru/2011/program">UC Forum</a> took place March 22-23, 2011 in Moscow, and was the first industry event of this kind in the Russian Federation. The Forum was very well organized and piggybacked on an well-established Call Center conference that has been running for 10+ years and provided great facilities, registration desk, audio-video support, etc. The venue <a href="http://www.radisson.ru/slavyanskayahotel-moscow">Radisson Slavyanskaya Hotel and Business Center</a> was excellent, and I did appreciate having the conference center, the hotel, and the restaurants under one roof when the weather outside is not quite spring-like. The event organizers would like to establish the UC Forum as an annual event and stay at the same location, so that over time participants have the option to gradually shift from call center sessions to UCF sessions. Having seen the struggles of many new industry events, I think this is a smart approach.</span><br />
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<span style="font-family: Arial;">Polycom's partner CROC Inc. had a <a href="http://www.flickr.com/photos/20518315@N00/5622042355/in/photostream">booth</a> showing Microsoft-Polycom integration, and a range of Polycom products. The booth was centrally located and quickly became a convenient meeting point.</span><br />
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<span style="font-family: Arial;">The conference was moderated by Denis Klementiev who did an excellent job introducing the speakers and managing questions from the audience. I counted about 80 people in the room (there were 150 registered participants but people are coming in for a particular session and then moving on). Almost all presentations, including my talk, were in Russian, and this put the audience at ease, led to many questions, side discussions, and introductions. </span><br />
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<span style="font-family: Arial;">My speaking slot was on the first day of the conference, and I always sit in the sessions before me so that I do not repeat things and can refer to information already covered by previous speakers. Here are the highlights. </span><br />
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<span style="font-family: Arial;">Mikhail Kochergin from Microsoft talked about the business case for UC. One low-hanging fruit is unified directories which eliminate the need to enter the same employee information in multiple directories (PBX, Email, Web, etc.), and save cost and time. Mikhail then focused on cost savings from teleworking (that seems to be very important for Moscow with its horrendous commute traffic) and from lower real estate cost (less office space). He also touched on some vertical applications such as telehealth where UC truly saves lives. It turns out that 8-9 people die every year in the Russian Federation while travelling to a medical facility; these and other lives could have been saved through telehealth applications. Mikhail analyzed how the major players approach UC, and stressed that Microsoft was focusing on ease-of-use and on allowing any device to access UC services. </span><br />
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<span style="font-family: Arial;">Stanislav Cherkov from CROC Inc. presented 5 case studies with Microsoft Lync and Exchange but also with audio and video equipment from Polycom. He talked about savings from IP telephony among distributed corporate offices across the Russian Federation and highlighted the tremendous traffic increase once the UC solutions were deployed. Most demand seems to be for integration of voice, instant messaging, presence, email, and calendaring but multipoint video is often required, as is integration with Avaya that has a strong position in the Russian voice communications market. </span><br />
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<span style="font-family: Arial;">Next was my presentation that focused on the global developments around UC, as well as on the standardization and interoperability work in international organizations. I included some UC market segmentation information and market forecast that shows robust growth of both 'Basic UC' that enables presence indicators to guide manual user selection of voice, email, or IM from a unified communications client and 'Enhanced UC' that augments basic UC by tying into business processes, supporting mobile workers, and seamlessly integrating videoconferencing to drive business differentiation. I covered the different deployment models – on-premise, hosted, and cloud-based – and focused on the <a href="http://www.broadsoft.com/products/broadcloud/">BroadCloud service</a> developed jointly by BroadSoft and Polycom. Finally, I provided a summary of the work in <a href="http://videonetworker.blogspot.com/2010/10/inside-unified-communications.html">UCIF</a>, <a href="http://www.imtc.org/">IMTC</a>, and other organizations with focus on interoperability. The global perspective was very well received and resulted in a lot of questions, so that the session ran over, and I had to "borrow" time from the next speaker: Pavel Teplov from Cisco (sorry!) </span><br />
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<span style="font-family: Arial;">The bottom line is that UC is impacting all areas of communication. Since no one vendor can address all UC areas, vendor ecosystems are gaining momentum, while standards and interoperability are becoming more critical … as are organizations such as UCIF that tests and certify interoperability. The presentation gave me the opportunity to reiterate Polycom's commitment to the Russian market, the agreement with <a href="http://www.pkcc.ru/content/pkcc.htm">РКСС</a> to manufacture Polycom equipment in the Russian Federation, and the opening of a new demonstration center in Moscow in fall 2011.</span><br />
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<span style="font-family: Arial;">I stayed for the rest of the UC Forum, and found all presentations very practical and informative. They provided a great overview of what is happening in the Russian Federation in terms of UC deployments. In particular, I enjoyed the presentation by Andrey German who is responsible for the video communications of the Superior Court of the Russian Federation. Apart from the fact that they are using a lot of Polycom equipment, I found the application very unique and compelling. It turns out the Russian Federation has a law that allows court proceedings to be conducted over video, if the court decides it is appropriate. That is very cost effective in a country that spans over 9 time zones (11 before President Dmitry Medvedev cut the number to 9 last year) and is in fact the largest country in the world - with 17 million square kilometers or 6.56 million square miles. </span><br />
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<span style="font-family: Arial;">The Russian trade press was in the audience and a lot of the questions came from journalists. I did a quick search today, and found several articles about the UC Forum, for example, by <a href="http://www.iksmedia.ru/news/3710336.html">IKS Media</a> and <a href="http://www.worldinfocomm.ru/4all/news/id/300/">World Info Comm</a>. Detailed description of the UC Forum in Russian language is <a href="http://www.ixbt.com/comm/ucf-2011.shtml">here</a>. </span>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-23278079757087909602011-04-06T05:34:00.001-07:002011-04-06T10:54:07.152-07:00What Did the 80th IETF Meeting Mean to HD Voice, HD Video, and Unified Communications?<span xmlns=""><span style="font-family: Arial, Helvetica, sans-serif;">IETF has been making Internet standards - called unpretentiously "Request For Comments" or "RFCs" but nevertheless working quite well – for exactly 25 years now. Happy birthday, IETF! May the next 25 be equally exciting! This anniversary also means that the Internet has matured, and I could feel it in the discussions at the 80<sup>th</sup> IETF meeting (</span><a href="http://www.ietf.org/meeting/80/"><span style="font-family: Arial, Helvetica, sans-serif;">IETF 80</span></a><span style="font-family: Arial, Helvetica, sans-serif;">) last week. </span><a href="http://www.flickr.com/photos/20518315@N00/5593076325/"><span style="font-family: Arial, Helvetica, sans-serif;">Prague</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> was a beautiful venue for the event and the meeting hotel </span><a href="http://www.flickr.com/photos/20518315@N00/5592936615/"><span style="font-family: Arial, Helvetica, sans-serif;">Hilton Prague</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> provided excellent facilities. </span><br />
<span style="font-family: Arial, Helvetica, sans-serif;">I do not attend most IETF meeting – that would be quite difficult with my busy schedule and with three IETF meetings on three different continents happening every year. The last one I attended (</span><a href="http://videonetworker.blogspot.com/2009/04/summary-of-74th-ietf-meeting-in-san.html"><span style="font-family: Arial, Helvetica, sans-serif;">IETF 74</span></a><span style="font-family: Arial, Helvetica, sans-serif;">) was in San Francisco two years ago, and I was very excited to find out what had changed in 2 years. The very strong </span><a href="http://www.flickr.com/photos/20518315@N00/5593519632/"><span style="font-family: Arial, Helvetica, sans-serif;">Polycom team</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> included Mary Barnes, a long-time IETFer, Stephen Botzko, who also covers ITU-T for Polycom, Mark Duckworth, and me. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">From the 133 or so session I counted in the program I attended the ones that were related to voice and video (over IP, of course, since it is all about the Internet), Unified Communications, and related technologies. I could recognize three main discussion topics: connecting IP communication islands, enabling UC applications on the network, and handling of multiple media streams.</span></span><br />
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<span style="font-family: Arial, Helvetica, sans-serif; text-decoration: underline;">Connecting IP Communications Islands</span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">IETF recognizes that there are still islands of IP communication and the vision of IP networks replacing the Public Switched Telephone Network (PSTN) is far from being fulfilled. This led to the </span><a href="http://www.ietf.org/proceedings/80/agenda/vipr.txt"><span style="font-family: Arial, Helvetica, sans-serif;">VIPR</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> activities (VIPR stands for "Verification Involving PSTN Reachability") that leverage nothing else but the PSTN to allow more voice to flow over IP and never cross PSTN. Since IP communication islands do not trust each other, the VIPR idea is to use a basic phone call to verify the destination is what it claims to be. On more generic level, the mechanism can be used to extend trust established in one network (e.g. PSTN) to another network (e.g. IP) but the VIPR working group seems to be focusing on the narrow and practical application of connecting voice over IP islands without PSTN gateways. VIPR is very important to HD voice because it enables direct end-to-end HD voice connections. PSTN gateways on the other hand always take the voice quality down to "toll quality" (3.4kHz, G.711), even if handsets and conference servers support HD voice. VIPR can be used for video, and in fact is even more beneficial for video, since PSTN does not support video at all. Once PSTN is used to verify the destination, all subsequent calls between source and destination can be completed over IP. Again, the quality is not limited by any gateway, only by available bandwidth, and HD video can flow freely end-to-end. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">Another interesting discussion that reminded us – who mostly live in the IP world - about the existence of PSTN was about Q.850 error codes generated by switches in the PSTN network. I remember the discussions about mapping these error codes to SIP error codes from previous IETF meetings but it turns out these mappings do not work well because some of the Q.850 have no equivalent in SIP and inventing new error codes only complicates SIP and confuses SIP servers. So the proposal on the table is to update </span><a href="http://www.rfc-editor.org/rfc/rfc3326.txt"><span style="font-family: Arial, Helvetica, sans-serif;">RFC 3326</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> "The Reason Header Field in SIP" to transport the original Q.850 codes. Well, as they say, when mapping does not work, it is best to encapsulate and let network elements decide what to do with the information inside. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">And then there was the discussion about lost Quality of Service (a.k.a. </span><a href="http://www.ietf.org/rfc/rfc2474.txt"><span style="font-family: Arial, Helvetica, sans-serif;">DSCP</span></a><span style="font-family: Arial, Helvetica, sans-serif;">) settings when IP traffic passes through a service provider IP network. Since SPs do not really like any of their customers telling them how to prioritize traffic in their network, they basically resets the QoS values in the IP packets coming from the customer LANs to something they can use in the SP networks. The problem is that the destination IP LAN may want to honor the original QoS but the packets coming in from the SP do not have the real DSCP values. This leads to all sorts of creative ideas how to pass more granular information about the type of application end-to-end. One proposal in the </span><a href="http://www.ietf.org/proceedings/80/agenda/mmusic.htm"><span style="font-family: Arial, Helvetica, sans-serif;">MMUSIC</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> working group was to update </span><a href="http://www.ietf.org/rfc/rfc4598.txt"><span style="font-family: Arial, Helvetica, sans-serif;">RFC 4598</span></a><span style="font-family: Arial, Helvetica, sans-serif;">, so that the session description (Session Description Protocol, or SDP) has more detailed description of the application (for example telepresence, desktop video, personal video, web collaboration, etc.), so that the destination LAN knows what priority to assign to the traffic in that session. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">Finally, there is always the topic of end-to-end security. Many obscure mechanisms defined in some RFC and being use in some application somewhere in the world turned out to have problems with the relatively new ICE firewall traversal mechanism. ICE stands for Interactive Connectivity Establishment, and is finally an RFC (</span><a href="http://tools.ietf.org/html/rfc5245"><span style="font-family: Arial, Helvetica, sans-serif;">RFC 5245</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> to be precise). As a result, there were numerous presentations at IETF 80 – mostly in the MMUSIC working group - about things that do not work with ICE: some fax scenarios, the relatively new DCCP protocol (which was all the rage back at IETF 74), simulcast streaming scenarios, and media aggregating scenarios. At this point, however, no one is considering changing ICE and the pretty universal response to such contributions was "sorry but we cannot help you". Another security issue was discussed in the </span><a href="http://www.ietf.org/proceedings/80/agenda/xmpp.txt"><span style="font-family: Arial, Helvetica, sans-serif;">XMPP</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> group, where the consensus was that Transport Layer Security, or TLS, was not working at all on the interface between XMPP servers, that is, in XMPP federation scenarios. The proposal in discussion was to use DNS SEC to verify SRV records and define rules how authorizations and permissions are handled across domains. The simplest explanation is that if you have Gmail with 1000 domains and WebEx with another 1000 domains, trying to establish XMPP federations among all domains would drive the number of connections towards a million, which leads to scalability and performance issues. Instead, the folks in the XMPP group want to establish one connection between Google and WebEx and have all domains use it. There is also a certain time pressure to solve the secure XMPP federation issue because the US government is considering implementation of XMPP federation - if the security issues are fixed. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif; text-decoration: underline;">Enabling UC Applications</span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">The second big topic at IETF 80 was standards around UC functions. While there are many different opinions where UC will happen – in a soft client, in the mobile phone, in the browser, etc. - at IETF 80, the attention was on "UC in the browser". </span><a href="http://www.alvestrand.no/pipermail/rtc-web/2011-March/000545.html"><span style="font-family: Arial, Helvetica, sans-serif;">RTC Web</span></a><span style="font-family: Arial, Helvetica, sans-serif;">, or Real Time Communication on the World Wide Web, was feverishly discussed, starting with the Birds of Feather event on Tuesday (from the saying "Birds of a feather flock together", or an informal discussion group) and concluding with the first RTC Web working group meeting (although the group has not officially been established yet) on Friday. Web browsers today use incompatible plug-ins to communicate with each other and there is no interoperability across browser vendors. RTC Web's vision is a standard that allows real-time communication functionality (voice, video, and some associated data) to be exchanged across web browsers. The architecture is still fluid but it is clear that a group of companies, including Google and to a certain extent Skype, is interested in standardizing the media stream across browsers. As for using standard signaling (which basically means including SIP stack in the browser), there is no consensus, as browser vendors seem more comfortable with their own proprietary HTTP-based signaling, and promise gateways to SIP networks. I guess I understand why Google wants to do it all in the browser (this is the environment they can control more or less) but I am still puzzled by Skype's position – maybe they are willing to give up the Skype client for a strong play in the infrastructure. I am mostly interested in standards, so SIP in the browser sounds more interesting than proprietary protocols that then require gateways to connect to standards-based networks. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">Another interesting discussion around UC took place in the XMPP working group, and was about interaction between XMPP and SIP clients. Since UC in its basic form is presence, IM, voice, and video, and since XMPP is used for instant messaging (IM) and presence while SIP is used for voice and video over IP, I would say this particular discussion was really about UC (outside the browser, though). The Nokia team presented a proposal for interworking between XMPP and SIP based on a dual-stack (</span><a href="http://tools.ietf.org/agenda/80/slides/xmpp-5.pdf"><span style="font-family: Arial, Helvetica, sans-serif;">SIXPAC</span></a><span style="font-family: Arial, Helvetica, sans-serif;">). We at Polycom love dual-stack implementations – with most new video endpoints and even some high-end business media phones supporting SIP, H.323 and even XMPP while multipoint conferencing servers support many protocol stacks simultaneously. The proposed XMPP-SIP dual stack would make gateways between SIP and XMPP unnecessary, and would allow for richer user experience on the dual-stack client. The key benefit I see is if SIP URIs that are used to place a SIP call can be automatically resolved (by some server) into Jabber Identifiers (JIDs) used in XMPP, and vice versa. This would for example allow the user to check presence, start an IM session, and then seamlessly escalate it to voice and video. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif; text-decoration: underline;">Multiple Media Streams</span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">The third big topic at IETF 80 was handling of multiple streams, or as the IETFers say, multiple m lines in the session description. Applications for multiple streams range from multi-codec telepresence systems to video walls in situation rooms to just sending multiple media streams for redundancy. The most important of all is of course the CLUE activity. </span><a href="http://www.ietf.org/proceedings/80/clue.html"><span style="font-family: Arial, Helvetica, sans-serif;">CLUE</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> stands for "ControLling mUltiple streams for tElepresence" (I know – picking random letters to create a catchy group name is an art form that I do not appreciate enough), and the CLUE working group had its first meeting at IETF 80. </span><a href="http://www.flickr.com/photos/20518315@N00/5592934863/"><span style="font-family: Arial, Helvetica, sans-serif;">Mary Barnes</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> moderated the session. With about 80 people in the room, the discussion covered symmetric and asymmetric point-to-point and multipoint use cases. Encouraging was that many people in the room had read the use case description and 20+ people agreed to review and contribute. Later the group spent quite a lot of time going over assumptions (about 8 in total) and requirements (about 12 originally but one was later dropped). It became clear that CLUE will only focus on handling multiple media streams and leave architecture, signaling, etc. to other IETF groups. Most questions were about definition of terms such as "stream", "source", "endpoint", "middle box", "asymmetric", "heterogeneous", and "synchronization". The CLUE group will continue discussions at a virtual meeting in May and possibly a face-to-face meeting in June. Polycom's Mary Barnes in a chair of the CLUE working group, and will keep me updated on the progress. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">CLUE is very important because the video industry needs a consensus how to handle telepresence and other multi-stream applications. Since Polycom has announced support for the Telepresence Interoperability Protocol (</span><a href="http://www.imtc.org/activity_groups/tip.asp"><span style="font-family: Arial, Helvetica, sans-serif;">TIP</span></a><span style="font-family: Arial, Helvetica, sans-serif;">), I frequently get asked how CLUE relates to TIP. The short answer is "It does not". CLUE's charter is to develop standards for describing the spatial relationships between multiple audio and video streams and negotiation methods for SIP-based systems. This new work is completely separate from TIP and will support many use cases that TIP does not. The work was originally proposed by the IMTC Telepresence activity group (which Polycom also co-chairs), and was chartered by the IETF early this year. Participating companies include Polycom, Cisco, HP, Huawei, ZTE, and others. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">Note that ITU-T Study Group 16 is also active in that area but has a different charter, which includes multiple-stream signaling for H.32x systems, the creation of services (like accessibility), establishing requirements for telepresence media coding, telepresence control systems, and media quality recommendations for telepresence systems. ITU-T is planning to harmonize the H.32x multi-stream signaling with CLUE but, more importantly, the above mentioned companies are participating in both IETF and ITU-T, which is the best way to make sure the standards do not contradict. As far as the future of TIP goes, it is an interim solution for vendors to interoperate with Cisco telepresence systems. We will have to see how long it lasts in the market place - certainly once solutions are deployed they tend to stay for a while. As usual, various vendors will provide their own migration paths to the standard, for example, Polycom will continue to support TIP as long as it is necessary for our customers, and gradually migrate telepresence systems to CLUE - once the work in the CLUE working group is completed and the standard is ready. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">Back to the broader topic of handling multiple media streams. The topic came up in several presentations in the MMUSIC group. For example, Huawei proposed transmitting 3D video via multiple streams. Since the 3D image can be created through two 2D images – one for each eye - simulcast (i.e. sending Left and Right views in separate streams) can be used. Other options include frame packing (combine Left and Right views into a single stream) and 2D+auxiliary (synthesize Left and Right views from 2D video using auxiliary data such as depth and parallax maps). The draft introduces a new SDP attribute called "Parallax-Info" with parameters "position" and "parallax". While some IETFers expressed concerns about breaking the Real Time Protocol (</span><a href="http://www.ietf.org/rfc/rfc3550.txt"><span style="font-family: Arial, Helvetica, sans-serif;">RTP</span></a><span style="font-family: Arial, Helvetica, sans-serif;">), there are interesting elements in the draft and I will keep following it.</span><br />
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<span style="text-decoration: underline;"><span style="font-family: Arial, Helvetica, sans-serif;">Conclusion</span></span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">IETF 80 was a very productive meeting and a great gathering of technical experts from around the world. They all brought new ideas and very different areas of expertise: networking, voice, video, web, mobile, etc. The meeting provided an excellent opportunity for discussions </span><a href="http://www.flickr.com/photos/20518315@N00/5593521526/"><span style="font-family: Arial, Helvetica, sans-serif;">in</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> and </span><a href="http://www.flickr.com/photos/20518315@N00/5593523508/"><span style="font-family: Arial, Helvetica, sans-serif;">outside</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> the conference rooms. A lot of topics that I know from IETF 74 progressed quite well but did not disappear, just led to additional areas that require standardization. </span><br />
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<span style="font-family: Arial, Helvetica, sans-serif;">My takeaway from the meeting is that standardization work is like a marathon – it requires patience and persistence to get to the final line. The </span><a href="http://www.flickr.com/photos/20518315@N00/5593186437/"><span style="font-family: Arial, Helvetica, sans-serif;">Prague Marathon</span></a><span style="font-family: Arial, Helvetica, sans-serif;"> on Saturday was therefore a fitting metaphor and a great way to conclude a very productive, well-attended, and amazingly versatile IETF 80. (IETF participants were not required to run the marathon!) </span><br />
<span style="font-family: Arial;"></span></span>Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com1tag:blogger.com,1999:blog-6891359700208493876.post-81751915047695979712011-02-09T11:44:00.001-08:002011-02-09T11:46:42.362-08:00Video Interview at ITEXPOOn the last day of <a href="http://www.tmcnet.com/voip/conference/east-11/">ITEXPO East 2011</a>, I had a chance to sit with Erik Linask, Group Editorial Director at TMCnet, and talk about my experience during the ITEXPO event, about the key priorities for Polycom in 2011, and about our engagement in the Unified Communications Interoperability Forum.<br /><br /><a href="http://www.tmcnet.com/tmc/videos/video-review.aspx?vid=4053">The link to the 13-minutes video interview is here.</a> Let me know what you think!Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-47030457090364491642011-01-28T11:43:00.000-08:002011-01-28T11:45:52.634-08:00ITEXPO in Miami Next WeekMy next week will be very busy because ITEXPO and other industry events are running in parallel in Miami, Florida. I looked through the schedule and - as of today - I will be presenting in five sessions: three of them are part of <a href="http://www.tmcnet.com/voip/conference/east-11/attendees/e11-conferences.aspx">ITEXPO</a> while the other two are part of the <a href="http://www.ingate.com/SIP_Trunk_UC_Summit_Miami_2011.php">Ingate Summit</a>.<br /> <br />My first session at ITEXPO is on February 2 at 1:30pm local time and is titled <a href="http://www.tmcnet.com/voip/conference/east-11/attendees/e11-conferences.aspx?t=E#E-04">“How Does The Traditional Desktop Phone Fit Into The Evolving Enterprise User Experience?”</a> Frank Stinson from Intellicom Analytics is moderating and I will be sitting next to speakers representing other business phone makers. This session will explore how the trend to Unified Communications impacts the way people access voice services – through telephones but also through soft clients, smart phones, and tablet devices. How should desktop telephones evolve in this environment and how are desktop phone manufacturers planning to increase the value of their products given evolving user expectations?<br /><br />Then on February 3 at 1pm I will present in the session <a href="http://www.tmcnet.com/voip/conference/east-11/attendees/e11-conferences.aspx?t=C#C-02">“Making Telepresence Affordable and Reliable”</a> which will be moderated by TMC Executive Director Paula Bernier and will also include Matt Collier from LifeSize Communications. This session will discuss the perception that telepresence is expensive and will clog your network with HD video traffic. My talk will focus on the reduced network bandwidth consumption (i.e. minimizing or avoiding IP network upgrades) and on the new network architectures that allow for virtualization of conference resources shared within the entire organization or deployed by service providers to serve vast user communities.<br /><br />My last ITEXPO presentation is in the session <a href="http://www.tmcnet.com/voip/conference/east-11/attendees/e11-conferences.aspx?t=UC#UC-03">“UC Interoperability”</a> on February 3 at 2pm. David Yedwab from Partner Market Strategies and Analytics will moderate and there will be two other panelists: Alan Percy from AudioCodes and Allen Mendelshon from Avaya’s UC Strategy team. The session will address the need for interop and standards in multivendor environments and explore different aspects of multi-vendor UC interoperability.<br /><br />The Ingate Summit is organized in parallel to ITEXPO and I had the pleasure to present at previous Summits, the last one being in Los Angeles in October. This time, the Summit starts early with pre-conference service provider workshops and I will present in the session <a href="http://www.ingate.com/SIP_Trunk_UC_Summit_Miami_2011.php">"Generating Revenue from HD Video”</a> on February 1 at 5:30pm. Joel Maloff of Maloff NetResults is moderating, and I will share the time with Karl Stahl from Intertex Data. Since the audience is serive providers, I will focus on the managed and hosted telepresence services, and also address ITSPs with current hosted voice offering that would like to add HD video services without much CAPEX. I will also provide an update of the industry efforts in the areas of telepresence interoperability and B2B video communications.<br /><br />My second session at the Ingate Summit is the <a href="http://www.ingate.com/SIP_Trunk_UC_Summit_Miami_2011.php">"Town Hall Meeting: Unified Communications"</a> on February 3 at 9am. The list of panelists is fairly long: Chad Krantz from Brodvox, Dan York from VOIPSA, Karl Stahl from Intertex, Jeff Ridley from ShoreTel, David Yedwab from Market Strategy & Analytics Partners, and Gary Mading from Aastra. I will be wearing my UCIF hat in that session, i.e., representing the <a href="http://ucif.org/">Unified Communications Interoperability Forum</a>. In such large panel, there will be no time for slides but we will have a discussion around the scope of Unified Communications, and how different vendors approach UC. I will focus on the UCIF philosophy (certification, not standards development) and call for other companies to join and influence the discussions in UCIF.<br /><br />All in all, next week in Miami will be very hot - although they do tend to set the air conditioning in the Miami Beach Convention Center on “very cold”. I am sure there will be a lot of interesting discussions and I hope to meet some of the blog readers there in person. For the rest, I would recommend watching the video interview that I will give on the last day of the conference (February 4). I will summarize the news from the conference in my answers during this interview. Once the link to the recording becomes available, I will add it to this blog post.Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0tag:blogger.com,1999:blog-6891359700208493876.post-89868784069870021502011-01-24T14:30:00.000-08:002011-01-24T14:41:46.490-08:00Industry events, speaking engagements, and white papersIn addition to blog posts, Video Networker keeps a complete list of the industry events that I attend and the topics of my speaking engagements. It also has a section with links to my white papers. See below!Stefan Karapetkovhttp://www.blogger.com/profile/08183450844021421072noreply@blogger.com0