News from Industry

Can WebRTC save telephony?

bloggeek - Fri, 12/23/2016 - 12:30

There is no real peak telephony.

[Chad Hart is no stranger to my readers here. He runs webrtcHacks, part of the Kranky Geek team and works at Voxbone. This time, he takes a look at telephony and where it stands today – with and without WebRTC]

Back in April of 2015, I recall Google WebRTC Product Manager Serge LaChapell talking about the WebRTC team’s focus on mobile and how they wanted to kick “VoLTE’s butt”. To be fair he was referencing call connection times, but reading between the lines I like to believe he has had ambitions well beyond that – namely beating VoLTE and the traditional telephony network in minutes.

"We want to kick VoLTE's butt" @slac talking about #webrtc mobile improvements pic.twitter.com/f1aVnSbgMK

— Chad Hart (@chadwallacehart) April 15, 2015

For many years I have tried to keep track of how the traditional telecoms has fared against the emerging VoIP application world (what they sometimes derogatorily call “OTT”).  I have had two hypotheses for several years now:

  1. Traditional telecoms over the PSTN is past “peak” and will continue to decline
  2. Real time communications (RTC) in general has been on the decline but is poised to make a comeback thanks to better implementations and technologies like WebRTC

Let’s check the data to test these statements.

Peak Telephony? Maybe Not…

Digging into the statistics from various sources, I was surprised to find I was wrong about my first hypothesis on peak telephony.

The US market

Let’s start by taking a look at the situation in the US, one of the world’s largest communications markets. The Consumer Telecommunications Industry Association (CTIA) provides an annual update that sometimes includes Minutes of Use (MoU) and subscriber data. The data shows that on a subscriber-level, mobile telephony usage already peaked on a per-subscriber basis in 2007.

However, there is a growing number of data-only subscriptions for our tablets and other devices counted as subscribers. This negatively skews the numbers. Looking at “minutes” on a per capita basis is a cleaner metric, so let’s divide the minute figures by the US population. This shows a much more interesting picture where mobile phone usage for traditional calling actually went up by 16%.


Total US cellular telephony minutes appear to be rising after stalling for many years (
my calculations)

Checking the data against other FCC sources, this growth may be overstated but there is no clear evidence of decline. So what’s going on? Much of cellular’s continued volumes can be attributed to fixed-mobile substitution – both in terms of people dropping their fixed lines and as the FCC reports  “A significant percentage of homes with both landline and wireless phone access received all or almost all calls on wireless telephones despite also having a landline telephone.” If we assumed total PSTN calling was flat, then according to my estimates, a 30% annual decline in fixed line minutes would be required to explain the decrease. This is possible, but way faster than past usage declines in fixed so it is more likely cellular usage did indeed have a very good year in 2015.

There is no clear evidence of peak PSTN telephony in the US, so let’s check some other sources.

The UK market

The UK’s Ofcom is generally a much better datasource than the FCC since they look at communications as a whole within the UK and compare it to other countries.

They are a lot more pessimistic when it comes to PSTN-based telephony. Their data is very definitive showing a continued, gradual decline in PSTN call volumes going back to 2010.  With a -3% 5 year CAGR, no matter how you cut it, “operator” traffic is down.

Ofcom CMR 2016 report shows declines in operator voice usage

They have not released their global figures for 2015, but their 2014 report showed similar trends with mature markets declining (US, Western Europe, Japan & Korea). However, emerging marketings like China, India, and Russia show show growth and just make up for declines in the mature markets in 2014.

Does anyone care about the PSTN anyway?

Outside of adding touch tone dialing and going cordless, the Public Switched Telephone Network (PSTN) telephony user experience hasn’t exactly changed a whole lot in a hundred years. The PSTN is only one way to make calls – now we dedicated VoIP apps,  messenger apps with voice, and a growing number of video communications options. Do these new forms for RTC give us any hope of reversing traditional telephony’s demise? The data here is more positive.

Ofcom’s data shows an increasing usage of VoIP for voice calls and a very definitive increase in video call usage. This is consistent with their international research from a year earlier:

VoIP Apps Save the Day

So where do newer RTC apps and features fit into all this? Using Ofcom’s methodology, the 18 countries they track produce somewhere around 10 Trillion minutes a year. Microsoft has previously claimed Skype does up to 3 Billion minutes a day – that’s a Trillion minutes a year if one assumes around a 3 Billion daily average. Even if the true annual value is half of that, clearly Skype alone is meaningful compared to PSTN volumes.

Apple does not release any figures for its FaceTime service introduced in 2011, but presumably its usage is substantial, although less than Skype’s based on Ofcom’s past user surveys. WeChat, Line, and Viber all have more than 200 million monthly active users with various VoIP features. WhatsApp now has more than 1 billion MAU. Its VoIP calling feature launched in April 2015 has more than 100 million voice calls a day. Taken together, these other VoIP services are easily more than a trillion minutes a year.

At 10 to 20% of the PSTN’s volume, clearly VoIP traffic has a ways to go before it dominates the PSTN, but there is no doubt its volumes are meaningful in comparison.  Furthermore, these services are still growing. Certainly some of that growth will come at the expense of the PSTN, but it appears they are also encouraging more RTC use in general.

Does WebRTC matter?

WebRTC does not factor heavily into the services cited above, but that is poised to change. At only 5 years old, WebRTC has not had that much time to widely establish itself in relation to other VoIP technologies. Still, there are a few notable standouts – particularly notably Facebook Messenger. Facebook has stated it has more than 300 million monthly active users of Messenger’s VoIP features and just this week announced it had 245 million monthly video users. Other notable users include Snapchat and of course Google’s Hangouts and Duo services.

There are a lot of other WebRTC apps showing big user gains too such as Houseparty which reported it had 20 million minutes of usage a day last month – not bad for an app that only emerged from the ruins of Meerkat a few months ago. In addition, more traditional VoIP apps like Whatsapp and Skype are starting to use WebRTC, albeit in limited circumstances today but that will certainly grow too.

In aggregate, I estimate WebRTC-based services easily have over 500 million MAU this year across 2 billion devices. Comparing this to other VoIP technologies at the 5 year mark, WebRTC is way ahead. This bodes well for WebRTC to be an incremental driver of VoIP traffic and further accelerator of RTC.

Conclusions

I have been concerned that the desire of people communicate in real time reached its pinnacle long ago. Why focus on RTC if the trend is clearly toward “messaging” and other forms of textual interaction?  Has telephony peaked? The evidence suggests that is probably the case for the PSTN in developed markets, but there are plenty of pockets of growth. Where declines exist, they are gradual. Even better, there is a large body of evidence that VoIP services are more than making up the gaps of any declines and then some. This indicates that we are actually using real time communications more than even. The recent and rapid rise of many WebRTC services is a further shows that this trend is very likely to continue, or perhaps even accelerate. That’s great news for the hundreds of WebRTC vendors out there and those that have yet to come.

The post Can WebRTC save telephony? appeared first on BlogGeek.me.

Kamailio - MI Code To Be Removed

miconda - Tue, 12/20/2016 - 16:29
Management Interface (MI), the old line-based text protocol interface to interact with Kamailio is going to be removed in the near future. It was declared obsolete several years ago, when we introduced the RPC interface. With upcoming version 5.0 already having the code source tree restructured, this step makes a better and slimmer foundation for next generation of releases.The default configuration file is now shipping with jsonrpcs module and kamctl tool is using the RPC interface via jsonrpcs. The command line parameters for kamctl should be the same like for the past releases, but the output is now in jsonrpc format.The RPC interface is implemented by the following modules:
  • ctl – binary rpc protocol – with transport layers for FIFO file, unix sockets and IP sockets (both datagram and stream). It is the module used by the kamcmd tool.
  • jsonrpcs – jsonrpc protocol – with transport layers for FIFO file, unix and IP sockets (datagram) and HTTP via xhttp module. It is the module used by kamctl and kamcli tools.
  • xmlrpc – xmlrpc protocol – with transport layer for HTTP
If you were using mi_fifo or mi_datagram modules, then you can switch to jsonrpcs module (it offers the transports for the two mi modules). If you were using mi_xmlrpc, then you can switch to xmlrpc module.Thanks for flying Kamailio!

The Best WebRTC Security is Prone to the Stupidest Developer

bloggeek - Mon, 12/19/2016 - 12:00

WebRTC is the most secure technology for video communications. And yet – developers can screw this for you.

There is a rise in security breaches and data theft incidents in 2016. You see this from the amount of information out there. I’ve written about WebRTC and security for quite some time, but a recent post I’ve read compelled me to write about it again.

The post? Red Cross Blood Service in Australia leaks personal data.

The gist:

  • Site is secure
  • A contractor places database dump on the internet for backup
  • And that get found

It probably happens more often than not. You build a service. You take care of its security. And then, someone down the lines screws you over with his maintenance processes. To some extent, this is just as bad as social engineering, where a hacker tries to gain access by fooling people to believe he is someone else.

Make sure to download the WebRTC Security checklist. Print it and stick it on the wall behind your monitor so you don’t forget.WebRTC Security baseline

WebRTC comes with a few security concepts that are quite new and innovative in VoIP:

  1. In WebRTC, EVERYTHING is encrypted. Not only by default, but also in a way that can’t be modified – there is no way to send data over WebRTC in the clear
  2. WebRTC forces you to operate over HTTPS and WSS in your web application, so signaling gets encrypted as well
  3. Screensharing requires an additional layer of consent, be it whitelisting of your site or a creation of a browser extension
  4. Browsers today update frequently and automatically, so any security threat found gets patched faster than most enterprise and VoIP vendors react to their security breaches

The thing people forget is that WebRTC is just a piece of technology. A building block. It is up to the developers to decide how to use it in their own product. During that integration, security breaches can be created quite easily.

In the WebRTC course I launched two months ago, I’ve added a lesson dealing with WebRTC security. It goes through the mechanisms that exist in WebRTC and the areas that need to be further secured by the application.

Two big issues left to developers today are TURN passwords and access to backend server resources.

#1 – TURN passwords

TURN servers predate WebRTC. They are used by SIP (or at least are found in the spec), and there, the notion is that the user agent (=device/endpoint) is secure and “named”. So a username and password mechanism was created to get a TURN binding. The reason you want such a mechanism in the first place is because TURN servers are bandwidth hogs – they relay media, and by doing that they cost a lot in terms of bandwidth. So if you are paying for it, you don’t want others to piggyback on it.

The problem with this approach in WebRTC is that the username and password needs to be passed from your JavaScript code inside the browser to the server. Which means that information is available in the clear for many use cases – those where you don’t need or want the user to identify with the network at all.You also don’t want someone sniffing your code in the browser and then reusing these credentials elsewhere.

The current approach out there is to use temporary passwords (I like calling them ephemeral – it makes me sound intelligent). Ones that become useless in an hour or two.

This means that someone in your backend randomly creates a password that is short-lived and shares it with both the TURN server and the client.

The above illustrates how this is done.

  • The App Server, in charge of signaling in this case, creates a password. It updates the TURN server about said password and also gives that information to the User
  • The User then creates a peer connection, configuring the TURN server in it with the relevant temporary password

Great.

Now lets add a media server into the mix.

Who should be generating that password and passing it around to whom? Should the Media Server now be in charge of it, or is it up to the App Server still to take care of this?

Which leads me to the second important security aspect of WebRTC when it comes to your development – backend server resources you need to protect.

#2 – Backend server resources

In many cases, I find that when the work is outsourced, the end result tends to be a jumble of an architecture if things aren’t thought out properly from the beginning.

This usually causes the wrong servers to need to connect and communicate directly with the User. While not an issue on its own, it can easily turn into a headache:

  • Not having a clear picture of the state in your backend means you lose control – this can turn ugly when issues arise
  • Opening up more of your backend towards the internet means more points to secure against penetration
    • And yes – I know there’s a trend to treat servers in the cloud as if they are always open to the internet
    • Which means you need to think about how best to protect them in the first place anyway, which happens to be closing them as much as possible

What I suggest in many cases is:

  1. Media servers should never be controlled or accessed directly from the Internet
  2. Media servers should only pass media to and from the Internet
  3. Whenever they need to be controlled, you do that using backend-to-backend communication from other servers you have that are already managing the users on the Internet
What’s next?

I am not a security expert. I know a bit about it and try to stay informed, but I am by no means an expert in it.

You should make sure to take security into consideration when developing your service and don’t assume WebRTC does everything for you. It doesn’t, but it is the best starting point you’ll get.

If you want to learn more about WebRTC, I will be opening the course again for another round. Probably during April.

If you are a corporate looking to have an open access to course materials throughout the year for your workforce – I am going to announce such a plan soon, but feel free to reach out to me before that happens.

Just do me a favor – don’t leave WebRTC security to chance.

Need a reminder? Download the WebRTC Security checklist. Print it and stick it on the wall behind your monitor so you don’t forget.

The post The Best WebRTC Security is Prone to the Stupidest Developer appeared first on BlogGeek.me.

Slack Does WebRTC Video – Here’s How (Gustavo Garcia)

webrtchacks - Thu, 12/15/2016 - 21:53

Slack is an über popular and fast growing communications tool that has a ton of integrations with various WebRTC services. It also acquired a WebRTC company a year ago and launched its own audio conferencing service earlier this year which we analyzed here and here. Earlier this week they launched video. Does this work the same? Are […]

The post Slack Does WebRTC Video – Here’s How (Gustavo Garcia) appeared first on webrtcHacks.

How to Write the WebRTC Requirements for Your New Product?

bloggeek - Mon, 12/12/2016 - 12:00

Got a requirements document to write for WebRTC? Here’s a step by step guide to doing just that.

Here is something that I do with my customers quite often. In many cases, when I consult vendors, they are in the process of building a new product or integrating an existing product with some new communication capabilities. This involves using WebRTC and outsourcing the actual development.

More often than not, I find myself writing the baseline of the requirements document for the customer, to server as a WebRTC RFP (Request For Proposals) that get used to communicate the requirements with the potential outsourcing vendors.

I wanted to share the process that I use in writing the first draft of this document. To make this a bit more useful, let’s assume that what we want to do is build a webinars service, where a few people can join as the speakers in the webinar and people online can “listen in”.

I’ve created a WebRTC requirements template and a sample webinars requirements document that you can use when you need to write the requirements for your own product.

Get the WebRTC Requirements Template and Sample Webinars Requirements Document

Here’s step by step how I’d go about doing that.

Step #1: Structure your document

First things first. To make sure I don’t forget anything, I like to split my requirements document into 4 sections:

  1. Overview
  2. Architecture
  3. Functional
  4. Non-functional

As you can see in above, I place TBD for each section in the document. I do that for all sub sections that I add to the document as well. This way, I can easily search the areas that haven’t been filled in properly yet when I work on it. Most often than not, writing these WebRTC requirements take a couple of hours and span a few days because they are collaborative in nature.

I tend to leave out the mechanics around the project – such as the price model I am looking for, or the timeline of the project. These tend to change between companies and they often better reflected elsewhere than in the technical requirements that I try to describe here.

Step #2: Write the overview

First thing I do once I have the template ready for my needs is write the overview part.

I try to keep the overview short and sweet, with a focus on making sure people understand what it is that I am trying to achieve in the service – what my challenges are and what I consider as success.

Usually, 2-3 paragraphs should be enough.

Step #3: Describe the architecture

Now it is time to start thinking about our architecture. By that, I don’t mean the architecture of the solution, what processes, servers and switches I want – I leave that for the vendor to fill in. What I mean is the entities I have in my service, trying to focus around the session – the types of media and signaling I want running there.

I do this by going analog, and just jot it down on my whiteboard and taking a picture of the end result. I find this more natural for me than using Powerpoint or Vizio. Later on, I might redo it as a Powerpoint diagram, but more often than not, I just leave it as is.

Above is the drawing I just did to describe the BlogGeek.me Webinar I just invented.

After the visual, I explain the different entities that are in the drawing and the relations between them. This part is really important, as oftentimes, it will reveal entities or flows that I haven’t thought about earlier.

In the case of the BlogGeek.me Webinar, we’ve got multiple potential Speakers who interact using audio and video with each other in the Webinar, which then gets sent to multiple Viewers and also to an external Storage.

I try to keep things focused and to the bare minimum that is necessary for the understanding of the service.

Step #4: Fill in the features

To some extent, this step is the main chunk of what the product does. For me, this is a brain dump of the things a user should be able to do in the system.

There are different types of features you might be needing. I focus on those that relate to the communication part of the product and nothing else.

Here’s a checklist of what I usually go through when doing this:

  • Is this a 1:1 service or multiparty?
  • Audio? Video?
  • Any screen sharing or other collaboration capabilities?
  • Messaging/chat?
  • How do users get authorized, authenticated and connected?
  • Any in-session controls users need to have?
  • Any indicators to show on their display?
  • Do we need recording capabilities?
  • Anything that we missed?

Make sure you answer all the questions above as requirements in the document if they make sense and add your own to the list.

Here are a few of the ones I’ve written for the webinars product:

Notice how I’ve indicated that connectivity via PSTN is optional in a future phase? This serves two purposes for me:

  1. It gives the vendor a hint of what architecture to put in place to support this later on down the road
  2. It also gives the vendor a feeling that this is a journey and not a one-off project. He will be more committed to its success if he knows you might call on him later on to improve and extend the service
Step #5: Handle the non-functional requirements

Now it is time to go over the non-functional requirements. These are the boring and ugly details that can make or break a service, so spend enough time on this one.

What do I mean by non-functional? These will usually be things you will take for granted, but the vendor won’t. To reduce friction and arguments in the future, I add these. In most likelihood, if you don’t write these down, a vendor will ask about a few of these things anyway – so just write them down to reduce the unnecessary round trips and to make sure you and the vendor are on the same page.

I tend to split this section to 5 subsections, each with its own focus:

1. Devices

Here I list all the devices I want to support. Browsers, operating systems, mobile devices, etc.

Each gets its own special treatment. Things I usually look at here are:

  • Which browsers to support natively?
  • Do I need an Electron PC app? If I do, then on which operating systems?
  • What versions of iOS to support? Which earliest devices?
  • What versions of Android API to support? How many specific devices do I want tested?

In many ways, I derive the requirements here based on the WebRTC Device Cheat Sheet that I published.

2. NAT traversal

NAT traversal is often overlooked. There are two areas where I cover NAT traversal – here and in the Security subsection below.

Here, I define who takes care of it – do I expect the vendor to bring a NAT traversal solution, will I be doing it, or should they use a third party hosted service (there are a few out there offering it).

The second part that I sometimes decide here, but not always, is where I want it deployed – along with the media servers or closer to the connecting user. It is a matter of architecture needs that I prefer leaving to the vendor to fill in but not always (can’t really say when in a definitive way).

In my webinar example, I decided to make things easy and just use a third-party hosting service:

3. Scalability

For scalability I make sure I cover a few areas:

  1. What’s the scale of a single session? Just make sure that if there are different types of users, you indicate how each can scale
  2. What’s the scale of the service as a whole? How many sessions can exist concurrently?
  3. Do I need to address any geographical locations when it comes to scaling?
  4. How are the different parts of the systems scale? Independently of each other?

Here’s how I fit it into our webinar example:

4. Security

The security part is slightly tricky. First, because I am not an expert. But also because almost nobody is.

What I usually place here is the basics of how I’d like to see the backend (encryption between the servers), but I do cover two important areas:

  1. Media servers. I prefer access to them to be limited to the application server only when it comes to control and have all signaling be routed through the application server or a signaling server. I don’t like giving access to my resource hog openly over the internet. Call me old fashioned
  2. TURN servers. Here I always state that I want ephemeral passwords. Otherwise, vendors usually do the short route of using a static username and password, which in WebRTC is like no password at all on the TURN server
5. DevOps

The DevOps section deals with things required to run this product on a daily basis. I tend to fill in three main things here:

  1. Hosting – is this planned to be deployed on premise? In a specific cloud provider? Using Docker or some other container technology?
  2. Reporting – what type of information do you want to collect to generate reports on the use of the system? These should be for offline use – think a daily email or something similar
  3. Events and statistics collection – this is what you want to be able to collect to monitor the health of the service in realtime
Step #6: Do a one-over and share

Now that we’ve written t all, time to go over the whole document to make sure things aren’t missing:

  1. Clean up any leftover TBDs
  2. Add clarifications where necessary
  3. Add things not in the template

Here’s what I decided to add to the webinar example:

As you can see, for me, open source was really important.

Now that you are done – go share the document with your colleagues, and once approved internally, it is time to share with potential outsourcing vendors.

Why so short?

To some, this approach may seem a bit shallow. It doesn’t include all corner cases or describe in a lot of detail what goes on. The thing is, that there is a balance between what you can effectively do and achieve as a small startup or even a big company with a new project than what you’d do on a long running multi-year millions of dollars project.

For me, this proves itself as a good way to capture the essence of what it is that needs to be developed and getting replies from potential vendors to building the product. Once I get the replies, it is time to go over them and see who makes the most sense – a lot based on how they replied to the RFP in the first place.

What’s next?

So here’s how you should write your next WebRTC requirements document:

Step #1: Structure the document to make sure all bases are covered

Step #2: Focus on the overview – explain what your product needs to achieve

Step #3: Draw the architecture and explain it

Step #4: Write down your functional requirements

Step #5: Write down all non-functional requirements

Step #6: Do a one-over to make sure you didn’t miss anything

I’ve built a WebRTC Requirements Template document for you. You can copy it and fill it in with the requirements of your own product. It already holds many of the questions you’ll need to answer, so it can serve as a guide for you.

Now, to write this article, I also had to create a real-world example (remember our webinar service?). This example is also shared so you can see how I write things down.

Get the WebRTC Requirements Template and Sample Webinars Requirements Document

Oh, and if you still need help – I do offer a consulting service, where a lot of the time invested is placed into writing these requirements documents, finding suitable potential vendors and going over their responses.

The post How to Write the WebRTC Requirements for Your New Product? appeared first on BlogGeek.me.

Kamailio - Source Code Tree Restructured

miconda - Wed, 12/07/2016 - 16:28
Today – Dec 7, 2016 – the source code tree of Kamailio project was restructured into a slim and clean root folder. This was done in order to better handle various components and make it easier to get into the source. You can browse it online at:Here is a summary with the new locations:
  • src/main.c – the main c file of kamailio application
  • src/core/ – contains the source code for the core, including the subfolders for sip parser, memory manager and other core components
  • src/lib/ – contains the internal libraries
  • src/modules/ – contains the modules
  • utils/ – contains the tools used to operate kamailio such as kamctl and kamcmd
  • etc/ – configuration files
  • doc/ – resources for building documentations and core tutorials
  • pkg/ – packaging specs
  • misc/ – contains related resources, such as config examples, additional tools and scripts, vim syntax highlighting files, …
  • test/ – components related to testing
The Makefiles for building the application are in src/, with a new root folder Makefile that does target forwarding to src/. You should be able to use the same make commands inside root folder as well as inside src/.The kamailio binary is built in src/ directory.For example, next are the commands to build and run kamailio from source code tree, with debug mode logging to terminal:git clone https://github.com/kamailio/kamailio
cd kamailio
make all
./src/kamailio -f etc/kamailio.cfg -L src/modules/ -E -e -dddWhile some small adjustments may still be done, we hope that the new structure is going to make the long term management and development of the project smoother.Thank you for flying Kamailio!

WebRTC and Education – the Webinar Edition (and a Bonus)

bloggeek - Mon, 12/05/2016 - 12:00

Want to learn more about WebRTC in education?

Next week, testRTC will be hosting a webinar titled How WebRTC ushers the next wave of e-Learning innovation. As a co-founder of testRTC, I am tasked with the actual creation and hosting of the webinar, which means I will be speaking about what vendors are doing WebRTC when it comes to education and where I see their challenges.

I haven’t done a webinar in quite some time, so this is going to be fun for me.

We’ve decided to use Crowdcast as our webinars platform for it. Partially because it is a WebRTC based service, and I do love dog fooding. But also because I received some good reviews about it.

If I had to pick two very active verticals in the domain of WebRTC, these would be healthcare and education. We see this also at testRTC, where we help these vendors in testing and deploying their services to production.

Next Wednesday

So here’s what we’re doing next Wednesday – me and you:

  • On December 14 at 14:30 EDT, we’re going to meet online
  • I am going to give a few interesting examples of how education looks like when it meets WebRTC
  • Then talk about some of the challenges involved
  • You will have time to ask questions. I’ll answer them to the best of my ability
  • And then I am going to give you a bonus
The examples

The examples part of the webinar is probably going to be the most interesting one.

I remember talking almost 3 years ago with a startup in India about their use case. It was related to education and it blew my mind. It was so starkly different than what I assumed a startup in India would do within their local market for education that I saw it as my own private lesson. Since then, I talked with tens of vendors in this space. Each doing his own thing. Each focusing on solving a problem in tutoring. They are so wide in variety that you can’t even look at them as a single market.

But this is exactly what we will try to do here. I am going to categorize them a bit – I wonder where you will find yourself in that categorization.

The challenges

Learning has its challenges for the student, the teacher and now also for the platform.

My intent is to look at the challenges of the platform – what are the things necessary to put these different education systems in production and how to make sure they work properly.

For the various types of education platforms, I’ll give you tips for where you should focus with your testing – what are the weak spots to look for – so you can find and deal with them before your customers do.

The bonus

I am not going to say what the bonus is now – it will ruin the surprise. I will say though, that this is something you’ll find immediately useful.

The bonus will be available only to those who will be with me during the webinar itself, so register now and save your place.

What’s next?Register of course!

And feel free to write down your questions in advance – Crowdcast allows for that.

The post WebRTC and Education – the Webinar Edition (and a Bonus) appeared first on BlogGeek.me.

WebRTC, TURN and Geolocation. How to Pick the Best Server to Work With?

bloggeek - Tue, 11/29/2016 - 12:00

Different ways to do the same thing.

One of the biggest problems is choice. We don’t like having choice. Really. The less options you have in front of you the easier it is to choose. The more options we have – the less inclined we are to make a decision. It might be this thing called FOMO – Fear Of Missing Out, or the fact that we don’t want to make a decision without having the whole information – something that is impossible to achieve anyway, or it might be just the fear of committing to something – commitment means owning the decision and its ramifications.

WebRTC comes with a huge set of options to select from if you are a developer. Heck – even as a user of this technology I can no longer say what service I am using:

  • I use Drum for my Virtual Coffee sessions (haven’t done one in some time. Should do one next month)
  • I now use Jitsi meet for my Office Hours
  • Google Hangouts for testRTC meetings with customers
  • Whatever a customer wants for my own consultation meetings, which varies between Hangouts, Skype, appear.in, talky, GoToMeeting, WebEx, … or the customer’s own service

In my online course, there’s a lesson discussing NAT traversal. One of the things I share there is the need to place the TURN server as close as possible to the edge – to the user with his WebRTC client. Last week, in one of my Office Hour sessions, a question was raised – how do you make that decision. And the answer isn’t clear cut. There are… a few options.

My guess is that in most cases, the idea or thought of taking a problem and scaling it out seems daunting. Taking that same scale out problem and spreading it across the globe to offer lower latency and geolocation support might seem paralyzing. At the end of the day, though it isn’t that complex to get a decent solution going.

The idea is you’ve got a user that runs on a browser or a mobile device. He is trying to reach out to your infrastructure (to another person probably, but still – through your infrastructure). And since your infrastructure is spread all over the globe, you want him to get the closest box to him.

How do we know what’s closest? Here are two ways I’ve seen this go down with WebRTC based services:

Via DNS

When your browser tries to reach out the server – be it the STUN or TURN server, the signaling server, or whatever – he ends up using DNS in most cases (you better use DNS than an IP address for these things in production – you are aware of it – right?).

Since the DNS knows where the request originated, it can make an informed decision as to which IP address to give back to the browser. That informed decision is done in the infrastructure side but by the DNS itself.

One of the popular services for it is AWS Route 53. From their website:

Amazon Route 53 Traffic Flow makes it easy for you to manage traffic globally through a variety of routing types, including Latency Based Routing, Geo DNS, and Weighted Round Robin.

This means you can put a policy in place so that the Route 53 DNS will simply route the incoming request to a server based on its location (Latency Based Routing, Geo DNS) or based on load balancing (Weighted Round Robin).

appear.in, for example, is making use of route53.

Amazon Route 53 isn’t the only such service – there are others out there, and depending on the cloud provider you use and your needs, you may end up using something else.

Via Geo IP

Another option is to use a Geo IP type of a service. You give your public IP address – and get your location in return.

You can use this link for exampleto check out where you are. Here’s what I get:

A few things that immediately show up here:

  • Yes. I live in Israel
  • Yes. My ISP is Bezeq
  • Not really… Tel-Aviv isn’t a state. It is just a city
  • And I don’t live in Bat Yam. I live in Kiryat Ono – a 20km drive

That said, this is pretty close!

Now, this is a link, but you can also get this kind of a thing programmatically and there are vendors who offer just that. I’ve head the pleasure to use MaxMind’s GeoIP. It comes in two flavors:

  1. As a service – you shoot them an API and get geo IP related information, priced per query
  2. As a database – you download their database and query it locally

There’s a kind of a confidence level to such a service, as the reply you get might not be accurate at all. We had a customer complaining at testRTC servers which jinxed his geolocation feature and added latency. His geo IP service thought the machine was in Europe while in truth it was located in the US.

The interesting thing is, that different such services will give you different responses. Here’s where I am located base (see here):

As you can see, there’s a real debate as to my exact whereabouts. They all feel I live in Israel, but the city thing is rather spread – and none of them is exact in my case.

So.

There are many Geo IP services. They will differ in the results they give. And they are best used if you need an application level geolocation solution and a DNS one can’t be used directly.

Telemetry

When inside an app, or even from a browser when you ask permission, you can get better location information.

A mobile device has a GPS, so it will know the position of the device better than anything else most of the time. The browser can do something similar.

The problem with this type of location is that you need permission to use it, and asking for more permissions from the user means adding friction – decide if this is what you want to do or not.

What’s next?

I am sure the DNS option is similar in its accuracy level to the geo IP ones, though it might be a bit more up to date, or have some learning algorithm to handle latency based routing. At the end of the day, you should use one of these options and it doesn’t really matters which.

Assume that the solution you end up with isn’t bulletproof – it will work most of the times, but sometimes it may fail – in which case, latency will suffer a bit.

 

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

 

The post WebRTC, TURN and Geolocation. How to Pick the Best Server to Work With? appeared first on BlogGeek.me.

Siremis v4.3.0 Released

miconda - Mon, 11/28/2016 - 16:25
A new version of Siremis, the web management interface for Kamailio, has been released recently, respectively v4.3.0. The main work was towards making it compatible with PHP 7 and the new constraints from MySQL 5.7+. The announcement is available at:As a side note, the organizing of the 5th edition of Kamailio World Conference has started, being planned for May 8-10, 2017, in Berlin, Germany. More details will be available in the near future!Thanks for flying Kamailio!

Kranky Geek 2016 SF: Mobile WebRTC

bloggeek - Mon, 11/21/2016 - 12:00

Kranky Geek last week was quite a rush.

Wow.

What can I say. Last week, our Kranky Geek event was so much fun.

I won’t bore you with the details. We’ve focused this time on WebRTC in mobile. Got the best speakers possible – really. And had a blast of an event. I received so much positive feedback that it warms my heart.

I’d like to thank our sponsors for this event: Google, Vidyo, Twilio and TokBox. Without them, this event wouldn’t have been possible.

The videos are available online, and below you’ll find the playlist of the event:

Tomorrow we’re doing another Kranky Geek event. This time in Sao Paulo, Brazil. Different theme. Different sessions. I am dead tired, but working hard with Chad and Chris to make that a huge success as well. See you soon!

 

The post Kranky Geek 2016 SF: Mobile WebRTC appeared first on BlogGeek.me.

My WebRTC Device Cheat Sheet

bloggeek - Mon, 11/14/2016 - 12:00

All you wanted to know but didn’t know how to ask.

2 billion Chrome browsers? 7 billion WebRTC enabled devices by 2017? 50 billion IoT devices?

At the end of the day, who cares? What you are really interested in is to make sure that the WebRTC product you develop will end up working for YOUR target customers. If these customers end up running Windows XP with Internet Explorer 6 then you couldn’t care less about Apple, Safari and iOS support. But if what you are targeting is a mobile app, then which browser supports webRTC is less of an issue for you.

To make things a bit simpler for you, I decided to create a quick Cheat Sheet. A one pager to focus you better on where you need to invest with your WebRTC efforts.

This cheat sheet includes all the various devices and browsers, and more importantly, how to get WebRTC to work on them.

So why wait? Grab your copy of the cheat sheet by filling out this form:

  • Name* First Last
  • Company*
  • Email*
  • CommentsThis field is for validation purposes and should be left unchanged.
jQuery(document).bind('gform_post_render', function(event, formId, currentPage){if(formId == 9) {} } );jQuery(document).bind('gform_post_conditional_logic', function(event, formId, fields, isInit){} );jQuery(document).ready(function(){jQuery(document).trigger('gform_post_render', [9, 1]) } );

 

 

The post My WebRTC Device Cheat Sheet appeared first on BlogGeek.me.

Kamailio v4.4.4 Released

miconda - Wed, 11/09/2016 - 19:40
Kamailio SIP Server v4.4.4 stable is out – a minor release including fixes in code and documentation since v4.4.3. The configuration file and database schema compatibility is preserved.Kamailio v4.4.4 is based on the latest version of GIT branch 4.4, therefore those running previous 4.4.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v4.4.x.Resources for Kamailio version 4.4.4Source tarballs are available at:Detailed changelog:Download via GIT: # git clone git://git.kamailio.org/kamailio kamailio
# cd kamailio
# git checkout -b 4.4 origin/4.4Relevant notes, binaries and packages will be uploaded at:Modules’ documentation:What is new in 4.4.x release series is summarized in the announcement of v4.4.0:Thanks for flying Kamailio!

Desktop browsers support in WebRTC – a reality check

bloggeek - Mon, 11/07/2016 - 12:00

Time for a quick reality check when it comes to browsers and WebRTC.

I know you’ve been dying for Apple to support WebRTC in Safari. I am also aware that without WebRTC in your Microsft Internet Explorer 6 that you have deployed in your contact center there is no way for WebRTC to become ubiquitous or widely adopted. But hear me out please.

Browsers market share

The recent update by NetMarketShare on the desktop browsers market share is rather interesting:

It shows the trend between the various desktop browsers for the last year or so.

Here are some things that comes to mind immediately:

  • Google Chrome now has 55% market share. Its rise has stalled somewhat in the last couple of months
  • Microsoft Internet Explorer is still free falling. It will probably stop somewhere at 10% or so if you ask me
  • While Chrome gained the most users from Internet Explorer, it seems that Firefox has picked up users from Internet Explorer in the past two months
  • Microsoft Edge gained very little from the demise of Microsoft Internet Explorer. People who have adopted Windows 10 aren’t adopting Edge and are most probably opting to install and use Chrome or Firefox instead. I’ve mentioned it here in the past

What happens between Microsoft Edge and Apple Safari is even more interesting. Apple Safari is falling behind Microsoft Edge:

Something doesn’t add up here.

The Edge numbers should rise a lot higher, due to the successful upgrades we’ve seen for Windows 10 in the market. And it doesn’t. We already noticed how Chrome and to some extent Firefox enjoyed that switch to Windows 10.

I am not sure how the slip of Apple Safari market share from almost 5% in the beginning of this year to below 4% can be explained. Is it due to the slip in Mac sales in recent months or is it people who prefer using Chrome or Firefox on their Macs?

There’s one caveat here of course – these numbers are all statistics, and statistics do tend to lie. When going to specific countries, there will be a different spread across browsers, and to a similar extent, your service sees a different type of browser spread because your users are different. Here’s the stats from Google Analytics for this blog:

For me, it is titled towards browsers supporting WebRTC, and Safari is way higher than Edge and Internet Explorer put together.

Back to WebRTC

Every once in a while, someone would stand up and ask: “But what about Internet Explorer?” when I talk about WebRTC. It is becoming one of these questions I now expect.

Here’s what you need to think about and address:

  • Chrome is probably your go-to browser and the first one to support with your WebRTC product
  • Firefox comes next, and growing. So keep your tabs on it to see how it “performs” with your product
  • Edge. Useless for most. Add support to it if:
    • You do voice only (should work nicely), and you want that extra market share
    • You know for sure your users are on Edge
  • Internet Explorer. Ignore
    • Microsoft probably won’t invest in having WebRTC support in it, so don’t wait for them
    • Use a plugin or whatever if you must
  • Safari. Ignore for now. Nothing to do about it anyway
What’s next?

I am working on a quick cheat sheet for you. One which will enable you to make fast decisions for browser support. It will extend also into apps and mobile. Probably by next week.

Until then, if you plan on picking up browsers to support, think of your target audience first. Don’t come up with statements like “IE must be supported” or “Without Safari I can’t use this technology”. You are just hurting yourself this way.

 

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

 

The post Desktop browsers support in WebRTC – a reality check appeared first on BlogGeek.me.

Kamailio IRC Devel Meeting – Nov 2016

miconda - Wed, 11/02/2016 - 19:37
A new Kamailio IRC devel meeting to has beep proposed to discuss the current major issues and the plans for next Kamailio releaseThe target time frame is Nov 08-10, 2016 (Tuesday, Wednesday or Thursday).If many developers are not available, it can postponed it to another date in the near future (make proposals if that is the case for you).A wiki page has been created for it:Add there the topics that you want to be discussed and your availability.Thanks for flying Kamailio!

Snappy Kamailio – TADHack Global 2016 Winner

miconda - Tue, 11/01/2016 - 16:54
Coordinated by its founder Alan Quayle, with the help of many local teams across the world, TADHack Global 2016 edition was organized during October 14-16 in over 30 cities, counting over 2600 registrations that resulted in over 170 hacks. It is probably one of the largest hackathons recorded so far, maybe even the largest, anyhow, for sure in the telecom world.One of the cities involved in the hackfest was Berlin, the local event was hosted by VoIP Labs, being managed by Dennis P. Kersten. On a rainy weekend, a bunch of VoIP enthusiasts met, paired and started hacking with various telecom APIs offered by the event sponsors. The results were amazing, 4 completed hacks, all of them winning at least one prize, 3 of them being awarded TADHack global prizes — the details about all winners can be found at:One of the global winning hacks, awarded by Canonical/Ubuntu, was named “Snappy Kamailio” done by Daniel-Constantin Mierla, the co-founder of Kamailio project. The title of the hack is not related to the behaviour of the application, as one could imagine, but to the rather new packaging system known as ‘snap’. Here is the definition of a snap taken from the snapcraft.io:A snap is a fancy zip file containing an application together with its dependencies, and a description of how it should safely be run on your system, especially the different ways it should talk to other software.Most importantly snaps are designed to be secure, sandboxed, containerised applications isolated from the underlying system and from other applications. Snaps allow the safe installation of apps from any vendor on mission critical devices and desktops.Aiming to ease the deployment of applications across different Linux-based systems, wrapped with proper layers of security, the snaps concept look very promising.Daniel’s remarks on the TADHAck event and its outcome:“”I am glad that I could participate to the TADHack Global 2016, the local event in Berlin made it easier in a rather busy period of traveling, huge credits to Alan and Dennis for making it possible.As for the hack, what Canonical/Ubuntu was offering during the hackfest was a perfect fit for me – a Linux/Ubuntu cloud infrastructure to meet the needs of scaling deployments and RTC platforms. As one of core developers of Kamailio SIP server project, I wanted to do something using it, that, after all, can also be useful for our community.Cloud, virtualization, containers — all very hot concepts these days. But snaps target to be even slimmer, still avoiding annoying issues such as broken dependencies and different versions of libraries on different distributions. I heard about them, but never got the time to play with. TADHack global offered the chance that I didn’t want to miss. The hack-intense environment and discussions with other people around helped to clarify some doubts (hey Torsten, Dominik, Dennis).Once I started to build snaps and test them running, I realized that the sandboxed snap restricts some privileges that Kamailio uses when running on Linux, such as creating raw UDP sockets. With a bunch of patches after many try-and-errors, I was able to get the stock Kamailio from our github.com repositorybuilt and run as a snap.It felt that the participation to the TADHack was fruitful already. The prize announced few days later came as a very pleasant complement awarded by Canonical/Ubuntu.Now I am looking forward to get new versions of Kamailio snaps with a more specific target functionality, such as a load balancer, SIP registrar, a.s.o. Let’s see how far I can go till the TAD Summit, Nov 15-16, 2016, in Lisbon, Portugal, where I will participate and show the Snappy Kamailio and the evolution after the hackfest. If you are in telecom or real time communications looking for future transformations of the market, it’s an event you should definitely attend!””Next are the relevant resources for Snappy Kamailio.The spec files to build the Kamailio snap and some instructions are available at:The slides of the TADHack Global pitch for Snappy Kamailio:The video recording of the pitch:If you are interested in Kamailio snaps, join our development community on sr-dev mailing list.Thank you for flying Kamailio!

Get Ready for Kranky Geek San Francisco AND São Paulo

bloggeek - Mon, 10/31/2016 - 12:00

Kranky Geek is coming to town!

WebRTC is maturing. We’re 5 years into this roller coaster and it seems most companies have already understood that they need to use WebRTC in one way or another. To many, this is going to be an excruciatingly painful journey. They will need to change their business model, think differently about how they develop products and even rewrite their core values.

One of the reasons we decided to launch Kranky Geek over two years ago was to have a place where developers can teach developers about WebRTC. Somewhere that isn’t already “tainted” with the telecom views of the world – not because they are bad – just because WebRTC can accomplish so much more. What we are going to do next with WebRTC takes place in November and will happen in two separate locations:

Kranky Geek San Francisco

San Francisco is where Kranky Geek started and where I feel at home when it comes to this event. We will be doing our 3rd Kranky Geek event in San Francisco (and 4th in total).

It will take place on November 18, at Google’s office on Spear street.

Our focus this time around is going to be mobile. We’ve got sessions lined up that should cover most of the aspects related to WebRTC and mobile. Things like using React, cross platform development, video compression, specific aspects in iOS as well as specific aspects in Android related to WebRTC.

If you are into mobile development with real time communications – then this is an event you don’t want to pass up.

There is also a new attendance fee that was added – $10 that gets donated to Girl Develop It. You may notice we don’t have a woman speaker this time – it is hard to find women speakers in this domain, so if you are one or know one – make sure to let us know for our future events.

I’d like to thank our sponsors who made this thing possible:

  • Google – who brought us WebRTC in the first place and is instrumental to the success that is Kranky Geek
  • TokBox – sponsoring both the San Francisco and São Paulo events. They will share their experiences with mobile aspects of WebRTC related to Android
  • Twilio – sponsoring both the San Francisco and São Paulo events. Their session in San Francisco will cover WebRTC and the Internet of Things
  • Vidyo – a new sponsor that is joining the Kranky Geek family, and probably the best one suited to talk about real time video compression technologies that make sense in mobile devices
Kranky Geek São Paulo

This will be my first time in Brazil and also the first time we run Kranky Geek in Barzil. As with San Francisco, the event is hosted at Google’s office in São Paulo.

Our focus for São Paulo will be back to the basics of WebRTC. We are trying this time to fill in the gaps – share resources and insights that developers who use WebRTC in their daily activities need. This is why we have a few sessions that are targeted at debugging and troubleshooting WebRTC in this event.

Registration for the São Paulo event is free.

For the São Paulo event, we got the help of a few sponsors as well:

  • Google
  • TokBox – at São Paulo, TokBox will share with us how to deal with device and connectivity issues when it comes to WebRTC sessions
  • Twilio – will be looking at the makeup of a WebRTC service, as the browser implementation of WebRTC is the beginning of the journey only
  • WebRTC.ventures – who are sponsoring this event for the first time, will give the overview and introduction to WebRTC
  • Callstats.io – will explain what you can find in getstats() and how to use it
See you there

I have my own session to prepare for the upcoming Kranky Geek, along with a lot of work to make these two events our best yet. There are also changes and modifications that need to make their way to the website –  but rest assured – these events have great content lined up for you.

If you happen to be in the area, my suggestion is come to the event – it is the best place to learn and interact with people who know way better than I do what WebRTC is in and out.

And if you want to meet me – just contact me. I’ll be “in town” for an extra day or so.

See you all at Kranky Geek!

The post Get Ready for Kranky Geek San Francisco AND São Paulo appeared first on BlogGeek.me.

How to limit WebRTC bandwidth by modifying the SDP

webrtchacks - Wed, 10/26/2016 - 17:12

WebRTC 1.0 uses SDP for negotiating capabilities between parties.  While there are a growing number of objects coming to WebRTC to avoid this protocol from the 90’s , the reality is SDP will be with us for some time. If you want to do things like change codecs or adjust bandwidth limits, then you’re going to need to “munge” […]

The post How to limit WebRTC bandwidth by modifying the SDP appeared first on webrtcHacks.

FOSDEM 2017 – RTC DevRoom – CFP

miconda - Tue, 10/25/2016 - 16:51
FOSDEM is one of the world's premier meetings of free software developers,
with over five thousand people attending each year. FOSDEM 2017
takes place 4-5 February 2017 in Brussels, Belgium. https://fosdem.org

This email contains information about:
- Real-Time communications dev-room and lounge,
- speaking opportunities,
- volunteering in the dev-room and lounge,
- related events around FOSDEM, including the XMPP summit,
- social events (the legendary FOSDEM Beer Night and Saturday night dinners
provide endless networking opportunities),
- the Planet aggregation sites for RTC blogs

Call for participation - Real Time Communications (RTC)
=======================================================

The Real-Time dev-room and Real-Time lounge is about all things involving
real-time communication, including: XMPP, SIP, WebRTC, telephony,
mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room
is a successor to the previous XMPP and telephony dev-rooms.
We are looking for speakers for the dev-room and volunteers and
participants for the tables in the Real-Time lounge.

The dev-room is only on Saturday, 4 February 2017. The lounge will
be present for both days.

To discuss the dev-room and lounge, please join the FSFE-sponsored
Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc

To be kept aware of major developments in Free RTC, without being on the
discussion list, please join the Free-RTC Announce list:
http://lists.freertc.org/mailman/listinfo/announce

Speaking opportunities
----------------------

Note: if you used FOSDEM Pentabarf before, please use the same account/username

Real-Time Communications dev-room: deadline 23:59 UTC on 17 November.
Please use the Pentabarf system to submit a talk proposal for the
dev-room. On the "General" tab, please look for the "Track" option and
choose "Real-Time devroom". https://penta.fosdem.org/submission/FOSDEM17/

Other dev-rooms and lightning talks: some speakers may find their topic is
in the scope of more than one dev-room. It is encouraged to apply to more
than one dev-room and also consider proposing a lightning talk, but please
be kind enough to tell us if you do this by filling out the notes in the form.
You can find the full list of dev-rooms at
https://www.fosdem.org/2017/schedule/tracks/
and apply for a lightning talk at https://fosdem.org/submit

Main track: the deadline for main track presentations is 23:59 UTC
31 October. Leading developers in the Real-Time Communications
field are encouraged to consider submitting a presentation to
the main track at https://fosdem.org/submit

First-time speaking?
--------------------

FOSDEM dev-rooms are a welcoming environment for people who have never
given a talk before. Please feel free to contact the dev-room administrators
personally if you would like to ask any questions about it.

Submission guidelines
---------------------

The Pentabarf system will ask for many of the essential details. Please
remember to re-use your account from previous years if you have one.

In the "Submission notes", please tell us about:
- the purpose of your talk
- any other talk applications (dev-rooms, lightning talks, main track)
- availability constraints and special needs

You can use HTML and links in your bio, abstract and description.

If you maintain a blog, please consider providing us with the
URL of a feed with posts tagged for your RTC-related work.

We will be looking for relevance to the conference and dev-room themes,
presentations aimed at developers of free and open source software about
RTC-related topics.

Please feel free to suggest a duration between 20 minutes and 55 minutes
but note that the final decision on talk durations will be made by the
dev-room administrators. As the two previous dev-rooms have been
combined into one, we may decide to give shorter slots than in previous
years so that more speakers can participate.

Please note FOSDEM aims to record and live-stream all talks.
The CC-BY license is used.

Volunteers needed
=================

To make the dev-room and lounge run successfully, we are looking for
volunteers:

- FOSDEM provides video recording equipment and live streaming,
volunteers are needed to assist in this
- organizing one or more restaurant bookings (dependending upon number of
participants) for the evening of Saturday, 4 February
- participation in the Real-Time lounge
- helping attract sponsorship funds for the dev-room to pay for the
Saturday night dinner and any other expenses
- circulating this Call for Participation to other mailing lists

See the mailing list discussion for more details about volunteering:
https://lists.fsfe.org/pipermail/free-rtc/2016-October/000285.html

Related events - XMPP and RTC summits
=====================================

The XMPP Standards Foundation (XSF) has traditionally held a summit
in the days before FOSDEM. There is discussion about a similar
summit taking place on 2 and 3 February 2017
http://wiki.xmpp.org/web/Summit_21 - please join the mailing
list for details: http://mail.jabber.org/mailman/listinfo/summit

We are also considering a more general RTC or telephony summit,
potentially in collaboration with the XMPP summit.
Please join the Free-RTC mailing list and send an email if you would
be interested in participating, sponsoring or hosting such an event.

Social events and dinners
=========================

The traditional FOSDEM beer night occurs on Friday, 3 February.

On Saturday night, there are usually dinners associated with
each of the dev-rooms. Most restaurants in Brussels are not so
large so these dinners have space constraints and reservations are
essential. Please subscribe to the Free-RTC mailing list for
further details about the Saturday night dinner options and how
you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc

Spread the word and discuss
===========================

If you know of any mailing lists where this CfP would be relevant, please
forward this email. If this dev-room excites you, please blog or microblog
about it, especially if you are submitting a talk.

If you regularly blog about RTC topics, please send details about your
blog to the planet site administrators:

All projects http://planet.freertc.org planet@freertc.org

XMPP http://planet.jabber.org ralphm@ik.nu

SIP http://planet.sip5060.net planet@sip5060.net
(Español) http://planet.sip5060.net/es/ planet@sip5060.net

Please also link to the Planet sites from your own blog or web site as
this helps everybody in the free real-time communications community.

Contact
=======

For any private queries, contact us directly using the address
fosdem-rtc-admin@freertc.org and for any other queries please ask on
the Free-RTC mailing list:
https://lists.fsfe.org/mailman/listinfo/free-rtc

FUSECO Forum 2016

miconda - Mon, 10/24/2016 - 16:49
The 7th edition of FUSECO Forum, organized by Fraunhofer Fokus Institute, takes place between November 3-4, 2017, in Berlin, Germany:This year, the focus is on understanding application drivers and technology evolution towards softwarized 5G networks and the industrial internet revolution.This year the FUSECO Forum will be accompanied by 2 additional international events on 5G, all of them held during the same week in Berlin. Don´t miss any of them and get ready for 5G and the Industrial Internet of Things!

Quiet please – people are studying

bloggeek - Mon, 10/24/2016 - 12:00

No article today.

My course is launching today: Advanced WebRTC Architecture Course.

I’ve got some solid attendance for it, along with a good bulk of high quality material lined up.

Hopefully, this will be a success.

If you are taking the course – then good luck and please share your thoughts with me – I’ve built this course for you and I’d like you to benefit from it as much as possible.

If you aren’t taking it but still want to attend – feel free to enroll. I’ll be closing up course signups end of this week, with no clear indication if and when I’ll be running it next.

Now quiet please – there are people studying in here. Somewhere. Hopefully.

The post Quiet please – people are studying appeared first on BlogGeek.me.

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