News from Industry

S doesn’t stand for Simple in SFU.

What are you doing about your WebRTC SFU requirements?

I have noticed recently that more and more companies are attempting the creation of their own SFU. SFU stands for Selective Forwarding Unit, and it is by far the most popular and cost efficient architecture today for multiparty video with WebRTC. With it, all participants send their video to a single entity (usually in multiple resolutions/bitrates), and that single entity decides (selectively) how to route the incoming video to all the participants.

One such popular framework is the Jitsi Videobridge.

Up until today, an SFU for WebRTC was rather simplistic. You had VP8 to contend with as a developer but that’s about it. Life was good. You built your service and mostly whined about incompatibility between browsers.

Things have changed.

Here are a few things that you need to take into consideration when you either build your own WebRTC SFU or adopt/acquire one from others:

• Do you use VP8 or VP9 in your SFU?
  • Google is already adding VP9 to Chrome
  • How long will it take until it catches on for some use cases?
  • VP9 is a better codec, so why not use it?
• Can it support multiple codecs simultaneously?
  • Before the end of this year, we will have VP8, VP9 and H.264 available to us in browsers
  • Not all browsers will support them all
  • VP8 seems like the lowest common denominator at the moment
  • This may change to H.264 when Microsoft Edge and Chrome support it though
  • An SFU supporting only VP8 will start looking old pretty fast – and won’t work on Edge
  • Staying in H.264/VP8 land will not perform as well as VP9 in terms of perceived quality for the users
  • So it would be beneficial to be able to use whatever is available at the best possible quality
  • Which makes it a lot more complex for an SFU – more decisions to make with more data points to take into consideration
• Mobile
  • Mobile doesn’t like multiple, simultaneous video decoders
  • Especially not when this is hardware accelerated – not all smartphone hardware can work this way
  • For mobile devices, you just might want to select a single video stream to send it – or combine multiple video streams to a single one (which looks more like an MCU, but who cares?)
• Broadcast
  • In many new use cases, people want to have multiple participants chatting, but many more passively viewing
  • Can an SFU scale there? And if it can’t, what will you do instead?

 

Like any other technology, once you get down to it, there’s more to do than the proof of concept. Consider these aspects at the beginning of your project – and not when you need to seriously rethink your architecture.

 

Planning on introducing WebRTC to your existing service? Schedule your free strategy session with me now.

The post What are the Challenges of DIY your WebRTC SFU? appeared first on BlogGeek.me.

Do you really want to trust companies like Google or Facebook, or other large organizations with your private communications? They rely on employees to dig into and analyze information when a back door is opened to them. They claim to need to be able to sift through information to pinpoint nefarious activities or corporate opportunities, but that means they are subjecting safety to human limitations. And that door is now available for anyone else that finds a key. How many examples of “hackers” accessing private information such as credit card numbers or social security numbers do we need before we stop trusting external organizations to keep us safe. Who holds them accountable if not the user? Wouldn’t we be truly safer by not putting in a back door at all? Weakening encryption to allow one entity in means weakening security across the board. That is why many believe that “if you don’t pay for the service, you’re more than likely the product!” And, as of yet there is not a very reliable way for the average user to counter this idea and monitor proprietary software to ensure they aren’t selling you and your data. Check out what users on a Slashdot forum had to say about the U.K.’s encryption standard and why having a middle man is not a secure option.

Trailrunner7 posted, “The U.K. government’s standard for encrypted voice communications, which already is in use in intelligence and other sectors and could be mandated for use in critical infrastructure applications, is set up to enable easy key escrow, according to new research. The standard is known as Secure Chorus, which implements an encryption protocol called MIKEY-SAKKE. The protocol was designed by GCHQ, the U.K.’s signals intelligence agency, the equivalent in many ways to the National Security Agency in the United States. MIKEY-SAKKE is designed for voice and video encryption specifically, and is an extension of the MIKEY (Multimedia Internet Keying) protocol, which supports the use of EDH (Ephemeral Diffie Hellman) for key exchange.

‘MIKEY supports EDH but MIKEY-SAKKE works in a way much closer to email encryption. The initiator of a call generates key material, uses SAKKE to encrypt it to the other communication partner (responder), and sends this message to the responder during the set-up of the call. However, SAKKE does not require that the initiator discover the responder’s public key because it uses identity-based encryption (IBE),’ Dr. Steven Murdoch of University College London’s Department of Computer Science, wrote in a new analysis of the security of the Secure Chorus standard. ‘By design there is always a third party who generates and distributes the private keys for all users. This third party therefore always has the ability to decrypt conversations which are encrypted using these private keys,’ Murdoch said by email. He added that the design of Secure Chorus ‘is not an accident.'”

For more information click here: http://it.slashdot.org/story/16/01/19/2151215/uk-voice-crypto-standard-built-for-key-escrow-mass-surveillance

Last edition of Kamailio Development Workshop happened about three years ago. Several people asked about it from time to time, therefore I thought of trying to organize a new edition. Rather short term by now due to various constraints, so the plan is to do it again in Alicante, Spain, during February 15-16, 2016.More details are available at:

• http://www.kamailio.org/w/development-workshop/

Besides spending 2 days digging into Kamailio C code, such event is also good for discovering and sharing development resources among participants, as well as meeting people from community — at the previous editions we met for first time with Victor Seva, Vicente Hernando and Seudin Kasumovic, which in turn contributed a lot of Kamailio code afterwards. It will be also testing phase for upcoming major release v4.4.0, maybe we can plan some stress testing session on site.Right now the goal is to see if there is any interest in such event in order to nail down organizing it or not. Should anyone consider to participate, write an email to:registration [at] kamailio.orgThank you for flying Kamailio!

Last Thursday we had the first virtual w3c webrtc wg interim meeting. Once we sorted out a few technical details it went quite well!

Meeting Home Page:
https://www.w3.org/2011/04/webrtc/wiki/January_14_2016#Virtual_Interim

Nobody cares anymore.

Nobody cares if you are a silo or federated as long as you’ve got an API

It used to be important. Interoperability. Federation. Communication across networks, devices and vendors. All useless now.

We’ve got our lowest common denominator: IP, HTTP – the web. We have our point of federation/aggregation – they now call it the home screen of a smartphone.

People got all riled up on my blog last week something about Wire needing to federate – check the comments. My view? Federating wouldn’t move an inch in their user’s base needle.

Today’s openness and messaging is all about being the platform and enabling others to connect to you. How is that achieved? By way of APIs. And by this new stupid word – “Bots” – which most probably stands for automation.

Why is Bots stupid? Because it just means using the API in a certain fashion.

Back to Messaging.

Silo

If you have a service. What happens if it is a silo?

You gain users to it. Slowly or faster. Doesn’t matter that much (though it probably does to you).

One day you want to add more utility to the service – some stickiness – making sure people don’t leave. So you add features. But you understand at some point that going it alone won’t move you fast enough, so you open it up to external developers and services.

You publish an open API.

Federation

Federation you say? You make your service accessible to other networks by interacting with them using the same protocol?

Great.

But what does that give you exactly? Same set of features you have, give or take a feature or two. Same utility. No stickiness. No differentiation. Not enough.

So you again wish to add features, but getting out of the core you’ve federated just places you in the position of proprietary features. And again you’ll one day understand it isn’t enough. Faster “innovation” and growth are required on that front – you can’t cater all of your customers’ needs. So you… open up! Slap an API on top of your service.

No one cares

Two alternatives. Same end result.

Time to move on. Nothing here to see.

Just make sure you have a solid infrastructure – and an API on top of it for others to integrate with.

 

Planning on introducing WebRTC to your existing service? Schedule your free strategy session with me now.

The post Messaging. Federated? Silo? Does it Matter with an API and Bots? appeared first on BlogGeek.me.

The features this week include: the addition of profile logging, functionality, and default configurations for mod_amqp and display update support for Panasonic devices in mod_sofia. Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! This week we are talking about 3D printing! And, head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

• FS-8728 [mod_amqp] Adding logging profile and functionality and added default configurations
• FS-8735 [mod_sofia] Display update support for Panasonic devices

The following bugs were squashed:

• FS-8719 [mod_conference] Fixed a segfault caused by building without video support, but specifying video_mute_png variable for a conference member
• FS-8720 [core] Fixed a segmentation fault when switch_channel_str2cause is called
• FS-8731 [core] Fixed a crash when leg-b invite video in voice call
• FS-8734 [core] Cleaned up video jitter buffer by adding some formatting to the debugging logs so the text jumps around less and fixing sequence number rollover code to handle rollover better.
• FS-8713 [core] Fixed a crash on bad video rtp stream by pushing a patch to make the sizes match. This was the original intention since we want to preserve the packet as-is while in the jitter buffer
• FS-8736 [spandsp] Fixed missing MEMMOVE macro in spandsp autoconf
• FS-8721 [core] Fixed an eavesdrop memory leak caused by moving bug_remove_all after destroy where it’s more than safe to kill bugs indiscriminately
• FS-8673 [core] Fixed a core dump on playback after “Decode Codec is not initialized!” log message

Interesting how vendors define themselves.

Oftentimes, when you ask companies to define themselves, you get a complete shopping list. The end result is that you are left without an answer that you can use. Enter Twitter, with its razor sharp 140 character descriptions of the account.

Here’s what the WebRTC PaaS vendors I covered in my report (that weren’t taken off market) and how they define themselves:

#Cloud #Communications #WebRTC #API Start building on https://developers.apidaze.io Makers of @ottspott_co. Happy New YearApizee is a provider of real-time communications on webFollow us for mobile app development news, hackathons & events. Check out our community link below. For questions about your device tweet our experts @ATTCaresPlatform as a service and free APIs, SDKs to integrate text, audio, video chat, web conferencing and more into your websites and applications.Bit6 revolutionizes how developers integrate communications into their mobile & web applications. Download beta SDK for iOS now!WebRTC pioneer, real-time communication for mobile & web, customer & workforce contextual collaboration, in-app video & live assistance, visual self-serviceUnify delivers world leading collaboration & unified comms solutions. Talk to us about #NewWayToWork #NW2W #futureofwork Tweets by Sally ^SH & Jett ^JMMaking the future of digital communications services happen. #Mobile #Messaging #Telecom #Cloud #IoT #WebRTC #SS7 #Monetizing #MessagingSecurityPowerful, Intuitive #RealTimeCommunications for an all IP-World #WebRTC #OTT #mobile #cloud #KandyMobile #Disruptor50OnSIP is a leading provider of real-time communications services for businessesThe SDK that has everyone talking. Grab yours at http://developer.oovoo.com . Now with in-call messaging and filters too!Communications for Internet of Apps. Open, cloud based video, voice, data communications for enterprise, social, consumer apps across #WebRTC and mobile.Plivo is a Cloud API Platform and a Global Carrier Services Provider for Voice Calls and SMS. Sign up for a free trial today.Add video, voice, and messaging features to your app in minutes with #respokeShow What You See! Deliver a better experience by adding real-time interactions to your web or mobile app. #CX, #FieldService #Telehealth, #CustServ & moreUse the Sinch APIs to enhance your app with Voice, SMS, Verification, Video, and Instant Messaging.Skylink – WebRTC, audio, video, embeddable real-time communication.TokBox,a @tefdigital company, operates the #OpenTok Platform, making it quick and easy to integrate real- time communications into your websites and mobile appsTropo, now part of Cisco, is a cloud communications API that makes it easy to build voice & text messaging apps. Completely free to try, pay-as-you-grow pricingChanging communications forever by empowering software people to build the future of our modern communications apps. For support: @TwilioHelpCloud platform for real-time communication app development

A few interesting observations:

• Some don’t have an API specific Twitter account – rather a corporate wide account. This makes it hard for developers to understand they have an existing offering for developers
• Some explain their position in the organization and not what they do, which is somewhat sad
• Some definitely are pivoting

As these vendors are in the same place, there’s obviously a lot of shared use of words. I’ve taken the Twitter definitions above, removed unique words and placed them in a tag cloud – the bigger the word – the more appearances in makes:

Community is the derivation of communications, which makes a lot of appearances.

Most focus on Mobile.

To be expected.

WebRTC was less popular in the description. Refreshing and interesting.

Messaging isn’t high on the agenda of most platforms.

Some suggestions to the vendors are in order:

1. If you are a service within a larger organization, make sure to have a Twitter account dedicated to developers
2. Focus on the developer who reads the description and needs to understand what you can do for them, and not who you are within the company
3. Bring the same clarity that Twitter description forces you to all your marketing collateral

 

Planning on introducing WebRTC to your existing service? Schedule your free strategy session with me now.

The post WebRTC PaaS Vendors as Twitter One-Liners appeared first on BlogGeek.me.

2016 will be the year of video!

I heard that in 2005 I think. And then 2006. And then in almost every following year.

I used to work in a video conferencing company. So it really mattered.

When video did happen… it happened outside of the domain of enterprise video conferencing systems. And it continues to grow predominantly there.

But the thing is – video still is minuscule. Voice isn’t that interesting or important as it used to be either.

If I had to chart the use of our basic communications options these days, it would probably look like this:

I’d also say that the only reason video is almost as big as voice is due to WebRTC and the passing of time – It is easier today than ever before to implement and add video chat capabilities anywhere. And there are people who tend to do video calls instead of voice ones – because they can, but not because video is that critical, mystical part we’re often led to believe.

Video definitely has its place in the world and is extremely useful. I do most of my own business through video calling with clients all over the world. Most of them have never met me in person and are still happy to work with me. With voice, it would be slightly harder to achieve.

What ticked me to this topic was a piece on Ars Technica about the adoption percentage of IPv6 in 20 years (hint: the smallest 2 digit number). While the two things are different, video hasn’t fared much better and has been around for even longer.

Video will be a slow process, but the end result will never be the pervasiveness of voice or the current ubiquity and growth of messaging in all of its forms.

You still waiting for video to happen?

Planning on introducing WebRTC to your existing service? Schedule your free strategy session with me now.

Test and Monitor your WebRTC Service like a pro - check out how testRTC can improve your service' stability and performance.

The post 20XX will be the Year of Video! Not… and how is this related to IPv6? appeared first on BlogGeek.me.

The FreeSWITCH 1.6.6 release is here! This is a routine maintenance and security release and the resources are located here:

Release files are located here:

• Tarball: http://files.freeswitch.org/releases/freeswitch/freeswitch-1.6.6.tar.gz
• Packaging: https://freeswitch.org/confluence/display/FREESWITCH/Installation (Please Note the updated location for 1.6 packages)

New features that were added:

• FS-8401 [verto_communicator] Added Speaker selection in settings model and video page and fixed model to modal
• FS-8545 [verto_communicator] Improve controls for screen share, fixed a read lock regression, do not allow video floor on a member with a reservation id set, and add missing code to deal with screen share part
• FS-8616 [verto_communicator] A new menu for moderator, added gain buttons, and removed the 3-dot-button, moving its behavior to member-name div
• FS-8264 [verto_communicator][verto] Adapted the layout select to new response, added a separated menu in members list to set its reservation id, and added all the reservation IDs in the return of “list-videoLayouts” command
• FS-8293 [verto][mod_conference] Made sanity level based on 1080p and added a video-quality conference profile parameter for specifying the motion factor when calculating video bitrate, defaulting to 1.
• FS-8595 [mod_conference] Improve auto bitrate in personal canvas mode and do not let auto bitrate exceed native picture size
• FS-8543 [mod_conference] Improve mute handling on conference and WebRTC
• FS-8546 [mod_conference][mod_verto] Make original video demo backward compatible with livearray-json-status
• FS-8529 [mod_conference] Added video-floor to personal canvas mode
• FS-8549 [mod_http_cache] Add support for AWS_ACCESS_KEY_ID and AWS_SECRET_ACCESS_KEY environment variables in S3 profiles
• FS-8547 [core] Add error log into stats to log when quality impacting events begin and end
• FS-8568 [core] Allow building using system OpenSSL without EC support
• FS-8632 [core] Add origination_audio_mode originate variable with options for sendonly, recvonly or sendrecv
• FS-8559 [mod_shout] Add “mpga” to the list of supported extensions
• FS-8433 [mod_sofia] Allow hangup cause to be set inside redirect data

Improvements in build system, cross platform support, and packaging:

• FS-8592 [Windows] Fixed some simple compiler errors
• FS-8333 [build][Debian] Added mod_hiredis.deb
• FS-8152 [Debian] Make sure to package the image directories too
• FS-8576 [Debian] Fixed a package upgrade issue related to the fonts being installed in multiple packages
• FS-8723 [Debian] Adding a file extension to the package build logs
• FS-8614 [verto_communicator] Add Debian developers install script and update README.md to reference it
• FS-8578 [mod_verto] Fixed build error for missing __bswap_64 on osx
• FS-8293 [verto] Add quality level 0 to conference (default is 1) and fix some logic in auto bandwidth

The following bugs were squashed:

• FS-8537 [mod_lua] Fixed a segfault caused by passing nil to various lua functions
• FS-8527 [mod_conference] Do not send the video of last_video_floor_holder to video_floor_holder if the videos are related
• FS-8569 [mod_conference] Fixed undefined symbol conference_cdr_test_mflag
• FS-8574 [mod_conference] Fixed a read write lock issue
• FS-8053 [mod_conference][mod_sofia] Fix for WebRTC’s SDP containing a=sendonly for video, but the client still receiving the video stream
• FS-8589 [mod_conference] Fixed using conference playback with full-screen=true not working correctly
• FS-8354 [mod_conference] Fixed G722 audio issues with mod_conference caused by previous commit fab43547
• FS-8602 [mod_conference] Fixed conference not auto-generating layouts properly when callers with no camera are present
• FS-8615 [mod_conference] Fixed a crash when quickly changing layouts and setting reservation ids
• FS-8542 [verto_communicator] Fixed the tooltips of video controls
• FS-8603 [verto_communicator] Added device validation to prevent lost microphones after reset
• FS-8640 [verto_communicator] Don’t clear conference member reservation id on members that don’t have a reservation ID
• FS-8590 [verto_communicator] Fixed sending malformed vid-res-id command when changing layouts by treating no res-id the same as clear
• FS-8556 [mod_verto] Screen shares are not recoverable so do not try
• FS-8293 [mod_verto] Fixed some regressions where speed test caused excessive downlink bandwidth
• FS-8633 [mod_verto] Fix for the first verto to join a conference does not get “conference-livearray-join” event
• FS-8599 [verto] Removed a workaround for Mozilla that is no longer needed for video size
• FS-8553 [config] Include verto_contact into the dial-string in the samples
• FS-8566 [core] Fixed calls failing when put on hold in bypass media mode with inbound late negotiation set to false
• FS-8573 [core] Fixed one way audio after resuming from hold in bypass media mode and fixed a core dump on playback after “Decode Codec is not initialized!” log message
• FS-8575 [core] Fixed DTMF not being passed from a to b during rfc 2833 events
• FS-8612 [core] Fixed a rare IVR originated calls crash due to read codec leak
• FS-8625 [core] Fixed a segfault caused by an external incoming call from Google Voice.
• FS-8642 [core] Fixed CF_VIDEO_READY being set on non-video calls
• FS-8713 [core] Fixed a crash caused by read exceeding buffer
• FS-8716 [core] Fixed the recording offset delayed by a few seconds for rtmp stream
• FS-8677 [core] Fixed a crash (possible memory corruption) after codec change
• FS-8585 [mod_commands] Expanded {} and <> to [] for each dial string in group_call to allow for multiple device registrations for the same user
• FS-8582 [mod_httapi] Fixed a crashed caused by null URL being passed
• FS-8588 [mod_httapi] Fixed a crash found while fixing unreliable digit collection
• FS-8619 [mod_rayo] Reply with conflict stanza error if bind is attempted with duplicate JID. Improve error handling when ‘ready’ callback fails.
• FS-8708 [mod_rayo] Fixed the example configuration to map to correct DETECTED_TONE event from spandsp_start_tone_detect
• FS-8621 [mod_av] Fixed H264 HD1080P video quality issues
• FS-8631 [mod_db] Updated the regex to allow DSN to match the rest of FS code
• FS-8643 [mod_sofia] Fixed some memory leaks
• FS-8715 [mod_sofia] Make the oubound_proxy on the profile consistent with how we do the same thing on the gateway
• FS-8679 [mod_sofia] Fixed sofia sending call completed elsewhere if not disabled by the option ignore_completed_elsewhere
• FS-8711 [mod_skinny] Fixed a couple of possible memory leaks in mod_skinny packet reading code
• FS-8722 [mod_skinny] Remove nested redundant mutex that could cause a hang

This week we had a number of bug fixes and a change to the packaging build logs.

Improvements in build system, cross platform support, and packaging:

• FS-8723 [Debian] Adding a file extension to the package build logs

The following bugs were squashed:

• FS-8708 [mod_rayo] Fixed the example configuration to map to correct DETECTED_TONE event from spandsp_start_tone_detect
• FS-8711 [mod_skinny] Fixed a couple of possible memory leaks in mod_skinny packet reading code
• FS-8722 [mod_skinny] Remove nested redundant mutex that could cause a hang
• FS-8713 [core] Fixed a crash caused by read exceeding buffer
• FS-8716 [core] Fixed the recording offset delayed by a few seconds for rtmp stream
• FS-8677 [core] Fixed a crash (possible memory corruption) after codec change
• FS-8673 [core] Fixed a core dump on playback after “Decode Codec is not initialized!” log message
• FS-8715 [mod_sofia] Make the oubound_proxy on the profile consistent with how we do the same thing on the gateway
• FS-8679 [mod_sofia] Fixed sofia sending call completed elsewhere if not disabled by the option ignore_completed_elsewhere

The FreeSWITCH 1.4 branch had a couple of bug fixes back ported. And again, keep in mind that 1.4 is quickly moving toward end of life and won’t be supported any longer except for high level security issues.

The following bugs were squashed:

• FS-8708 [mod_rayo] Fixed the example configuration to map to correct DETECTED_TONE event from spandsp_start_tone_detect

 

Who needs to communicate in enterprises anyway?

Everyone.

Communication is… overrated

But do we really need to treat it as if it is the most critical piece of the enterprise world?

I use multiple systems to make my calls these days. They are WebRTC based or proprietary apps such as Skype, WebEx or GoToMeeting. I grumble when I have to use a proprietary system and install stuff on my laptop, but that’s life.

It was like that for me even when working for enterprises in the past – big and small. Somehow, you always need to have a “phone system” and be reachable. But other than that? I’d say “omnichannel” as a buzzword has stuck to the contact center but is just as important in unified communications.

But in Unified Communications, Omnichannel means something really different – it means that you can now reach out to people on lots of different channels and mediums – picking up the ones most suitable for the taks – which most often than not ends up being different than what the corporate IT has decided you should be using.

And you know what? I couldn’t be bothered with it.

The essence of Unified Communications is the here and now. Real time communications. If a minute passed, it is no longer interesting. It is lost.

Hangouts. Talky. A phone call (international or otherwise). Skype. Anything else.

Just pick one and lets meet.

—

Enterprise Messaging though is a different story.

It isn’t focused in the here and now, but rather in collecting data and making it accessible. It is about synchronizing teams and aligning them – asynchronously.

And “omnichannel” there? It means integrations with anything and everything that is enterprise software.

Which makes it the point of access for an employee to his daily life in the office.

It is a lot more sticky these days than unified communications.

—

Unified Communication is on another rebranding rampage. We used to call it “Convergence” a decade or two ago. And when that felt old, we started calling it Unified Communications. There are analysts that are now coining the term WCC – Workstream Communications and Collaboration. A mouthful that simply says Unified Communications need to look at the Enterprise Messaging space and copy it.

The end result will still be a focus on the here and now. And it will still be overrated.

 

[show promotion title=”strategy-session”]

The post Unified Communications is Overrated appeared first on BlogGeek.me.

 

Dave Horton discusses his open-source, node.js-based SIP load balancer for FreeSWITCH servers (http://davehorton.github.io/drachtio-…) which is currently deployed in several service provider networks.

The challenges in testing WebRTC services How do you view WebRTC? As a VoIP technology or an Internet one? There’s a huge divide between the two, especially when it comes to testing them. WebRTC sits in the intersection between them, making it doubly hard to grok – and test. Tsahi will outline in this session a few of the main challenges, with suggestions on how to fix them. He will also use the opportunity to show how testRTC.com addresses these challenges.

Hey FreeSWITCHers!
Last week FreeSWITCH turned ten!

It has been a long road,
With many, many lines of code.
The community has been there through thick and thin,
And witnessed all the hard work put in.
Long nights, no break,
But, a wonderful piece of software they did make.
Blood, sweat, tears, and no perks,
Would result in a switch that just works!

We had no idea they would be so merciless,
So, we apologize on behalf of Rachael from Cardholder services.
Convey your appreciation with some flair,
Show Tony how much you care,
By clicking on link below,
And, send him something as thanks for Verto.

http://goo.gl/3WCNnI

Challenges ahead.

A shy over a year ago, I wrote about 3 startups: Talko, Wire and Switch

All of them looked promising. All were using WebRTC.

In 2015, Switch had a meeting with $35 million, along with quite a few successful deployments in businesses big and small.

A month ago, Talko got acquired by Microsoft. I’ve interviewed here the Talko team in the past. Selling to Microsoft shows. Shutting the company. With little objections from customers. It all points to a single conclusion – Talko has been a failure when it comes to the business side of it. It probably had a solid technology – otherwise – why would Microsoft acquihire the team and fold it into Skype? I am sure Ray Ozzie and the team of Microsoft veterans in Talko added to this acquisition, but there was no other value in this transaction.

The Talko Team expresses it best on their updated homepage:

However, as engaged as many of you have been, the reality is that the broad-based success of communications apps tends to be binary: A small number of apps earn and achieve great viral growth, while most fall into some stable niche.

Talko didn’t grow fast enough or big enough. Clementine’s acquisition by Dropbox is similar. A communication solution geared towards team/group/enterprise communications gets acquired for its team with the service being left behind, never to be seen again.

And that’s in the less competitive domain of the enterprise. What will be with Wire? The third company I wrote about.

On Android, Wire reportedly has 100K-500K installs. Assuming iOS has twice as much (I am trying to be positive), that still falls way short of any of the messaging services we usually hear about – they are measured by 100’s of millions. Of active monthly users – not installs.

It is hard to see how Wire can change its abysmal future without a serious pivot or a drastic change in current market trends. Some will say this is a matter of a directory service and network effects. I think it is a matter of strategy and luck. Where Wire failed to attract the crowds, a different messaging service – Telegram, with 50M-100M installs on Android and a reported 60M monthly active users.

Wire was formed in 2012 and Telegram in 2013. So we can’t say Telegram had any head start here.

WebRTC makes it too easy to build and launch a communication service, which in turn, makes it hard to build a viable business with it. The role of product managers and people who need to think of the business case is more important than the technologists building the service when it comes to WebRTC. At the same time, finding good developers who grok WebRTC isn’t easy either.

2016 is going to be crucial for Wire.

What do you see for your initiative in 2016? Do you have a business case and route to market and money, or are you tinkering with the technology, assuming that if you build it they will come?

 

Planning on introducing WebRTC to your existing service? Schedule your free strategy session with me now.

The post Can Wire Succeed Where Talko Failed? appeared first on BlogGeek.me.

When it is the wrong metric to track.

Microsoft playing the number games with Edge adoption stats

44.5 billion.

minutes.

That’s how long people have been using Microsoft Edge “just last month”, according to Microsoft:

Over 44.5 billion minutes spent in Microsoft Edge across Windows 10 devices in just the last month.

That other number of 200 million monthly active devices using Windows 10 is much more impressive.

I am interested in Edge due to WebRTC and ORTC. It is one of the missing pieces of our puzzle to get adoption (or at least that’s what we’ve been told).

So how can 44.5 billion minutes can be so unimpressive?

Do the math.

Let’s assume only half of Windows 10 users make use of Microsoft Edge.

This gets us to an average of 445 minutes a month per user, placing it at less than 15 minutes a day (!)

How many of these minutes are spent with an idle browser? I got a laptop and a desktop open 24/7 with Chrome running on them. Even when I am engaged in other applications.

Microsoft decided to announce a largish number to hide the fact that Microsoft Edge isn’t really getting the love and adoption they expected, which is sad. I’ve used it a couple of times on my own Windows 10 laptop. It does what it is supposed to do and does it well, but that’s about it.

The challenge is migrating from Chrome. It stores my credentials to the many sites I visit, it has that nice search bar that often times just finds the URL I need without really searching (it automatically completes from its history), there are the few extensions I’ve got installed. All in all, it does the work. It is bloated and a memory hog, but the time when this mattered (a year or two ago) passed already, so there’s very little incentive for me to switch browsers.

Microsoft is killing Internet Explorer 8, 9 and 10 in the same day next week, pushing businesses into Internet Explorer 11 or Microsoft Edge. This might gain them a percentage or two more in adoption of Microsoft Edge – not nearly enough. Microsoft will probably announce end of life for Internet Explorer 11 in a year or two – the sooner the better if they want Microsoft Edge to grow.

What else can Microsoft do to improve its position? I don’t know. I don’t believe they can. The opportunity to surpass Google Chrome had come and gone. They will need to wait for the next opening when Google falters with Chrome or make something enticing enough for people to switch. It is sad, as Microsoft Edge is technically sound – it made browsers interesting again.

For WebRTC, Microsoft Edge still makes no difference at all. We’ve seen a few announcements of ORTC support by some vendors, but that’s about it. There’s no urgency in vendors to support it. The discussions are still about Internet Explorer when it comes to WebRTC.

Where does that leave us?

• Companies who waiting for Microsoft to adopt WebRTC will continue to wait
• Those who haven’t waited have made the correct choice – deal with what’s available and don’t wait for the forces that be to save you
• While Apple might get WebRTC properly, Microsoft hasn’t. Introducing ORTC into Internet Explorer is what the market needs, but it won’t happen by Microsoft
• Mobile is unaffected, as consumption there is done by apps, so browser adoption issues are irrelevant for most

 

Planning on introducing WebRTC to your existing service? Schedule your free strategy session with me now.

The post When is 44.5 Billion a Small Number? appeared first on BlogGeek.me.

Submission of presentation proposals for Kamailio World 2016 is open. Deadline for submission is February 18, 2016, notification of accepted proposals will be done latest on March 03, 2016.Be aware that interesting proposals can be accepted before the deadline, we plan to have two intermediate review sessions before February 18, 2016, announcing any accepted presentations immediately. Note also that at the previous edition there were more proposals than available slots and the 2016 edition celebrates 15 years of Kamailio project development, expecting a significant increase of interest for the event. Therefore it is recommended to send your proposal as soon as possible, do not wait till deadline.To submit the proposal, fill in the web form at:

• https://www.kamailioworld.com/k04/call-for-speakers/

The main topic of the conference is Real Time Communications, with the majority of the content being about Kamailio and other open source projects in the area. However, like for the past editions, we welcome very interesting presentations beyond those subjects.If you are interested to look at the agenda from previous edition, visit:

• http://conference.kamailio.com/k03/schedule/

Have a great 2016! Looking forward to meeting many of you at the next Kamailio World!

Continuing the tradition, many members of Kamailio project community will be at Fosdem 2016and we are planning again a social event on Saturday night (Jan 30, 2016). Among participants announced so far: Alexandr Dubovikov, Daniel-Constantin Mierla, Federico Cabiddu, Giacomo Vacca and Lorenzo Mangani.Should you want to join us for some drinks, food and interesting chats, announce yourself on sr-users mailing list (sr-users [at] lists.sip-router.org). As usual, each participants pays for himself or herself.There will be also couple of talks touching Kamailio on various tracks of Fosdem.